asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	b12f6bc358	chan_sip: Hangup transferer/transferee when transfer to Parking fails When performing a SIP transfer to a Park extension, if the Park fails, chan_sip will currently not hang up either the transferer or the transfer target. This results in the channels being orphaned with no thread to service frames, resulting in stuck channels. This patch immediately hangs up the two channels if a Park fails. (closes issue ASTERISK-22834) Reported by: rsw686 Tested by: rsw686 (closes issue ASTERISK-23047) Reported by: Tommy Thompson Tested by: Tommy Thomspon Review: https://reviewboard.asterisk.org/r/3107 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	9433c584d8	Add the missing part of r400140 When the patch to add retry-on-forbidden-response was committed, part of the patch for chan_sip was not committed which caused the feature to be entirely nonfunctional. This corrects the code in question. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874 ........ Merged revisions 405033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@405081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	788dccff42	Reset peer outboundproxy on sip.conf reload If you set a peer's outboundproxy and then removed it from the config, this would not get picked up in a config reload. This patch fixes that by resetting it in set_peer_defaults(). Closes ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/ ........ Merged revisions 403634 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@403635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	fe7c661e12	chan_sip: keep same local (from) tag for outgoing register requests For outbound register requests the tag on the From line was updated every 20 seconds prior to a successful registration and also once for each registration renewal. That behavior can possibly cause the registration to be denied because of the different tag, and is not aligned with the intention of RFC 3261 8.1.3.5 "... request constitutes a new transaction and SHOULD have the same value of the Call-ID, To, and From of the previous request...". This updates chan_sip to have a field to keep the local tag in the registration structure and use that tag for registration requests where the callid is also unchanged. (closes issue ASTERISK-12117) Reported by: Pawel Pierscionek Review: https://reviewboard.asterisk.org/r/2988/ ........ Merged revisions 402604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	1664326ba9	chan_sip: notify dialog info ignores presentation indicator in callerid The presentation indicator in a callerid (e.g. set by dialplan function Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies are generated during extension monitoring. Added a check to make sure the name and/or number presentations on the callee (remote identity) are set to allow. If they are restricted then "anonymous" is used instead. (closes issue AST-1175) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2976/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2d5debf45c	chan_sip: Fix RTCP port for SRFLX ICE candidates This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	9731431f08	chan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers" While looking at ASTERISK-22236, Walter Doekes pointed out that when running "sip show peers", the setting being displayed can be confusing. The display of "N" used to mean NAT (i.e. yes). The NAT setting has gone through many different changes resulting in the display of different characters to try and convey what the current setting is for 'Forcerport' (A for Auto and Forcerport is currently on, a for Auto but Forcerport is off, Y for yes, and N for no). During the initial code review to try and clarify these settings (especially since "N" no longer meant what it used to mean in prior versions of Asterisk), Mark Michelson suggested using the full space available to display the settings which helped to make the settings very clear. That was a great suggestion. Therefore, this patch does the following: * The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No. * A column for the 'Comedia' setting has been added. It too will display the setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No. * UPGRADE.txt has been updated to document this change. (closes issue ASTERISK-22728) Reported by: Walter Doekes Tested by: Michael L. Young Patches: asterisk-forcerport-display-clarification_v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2941 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	b0288cc8ad	chan_sip: Allow a sip peer to accept both AVP and AVPF calls Adapts the behaviour of avpf to only impact the format of outgoing calls. For inbound calls, both AVP and AVPF calls will be accepted regardless of the value of avpf in the configuration. (closes issue ASTERISK-22005) Reported by: Torrey Searle Patches: optional_avpf_trunk.patch uploaded by tsearle (license 5334) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	aa0fcfff6b	chan_sip: Fix an issue where an incompatible audio format may be added to SDP. If preferred codecs included any non-audio format the code would mistakenly add the audio format, even if it was not a joint capability with the remote side. (closes issue ASTERISK-21131) Reported by: nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470) ........ Merged revisions 401497 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	44826ac192	Remove Port Restriction When Checking For NAT When trying to determine if a peer is behind NAT, we should not be using the ports when comparing addresses. This patch removes the port from being checked and just useds the addresses now. (closes issue ASTERISK-22729) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-remove-using-port-for-nat-check.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2927/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	5ee5dccf51	Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag A condition was added in a commit to fix ASTERISK-21374, that, if the SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT flag to the dialog. This condition should not have been there since it assumed that if Asterisk is in an environment where NAT is involved, that the auto_* nat settings or force_rport setting would be on in the global settings. If the nat setting in the global setting is set to 'nat=no' and then turned on for peers (which is not quite the recommended way, although it is allowed) this flag is never copied to the dialog resulting in problems like, REGISTER replies going to the wrong port. This patch removes this conditional check and will now always use the peer's flag which by this point in the code the checks on whether the peer is behind NAT or not (if using auto_force_rport) have already been run. (closes issue ASTERISK-22236) Reported by: Filip Frank Tested by: Michael L. Young Patches: asterisk-2236-always-set-rport.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2919/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	ea53dd63be	Prevent chan_sip from sending duplicate BYEs. When a 200 OK for an initial INVITE is received, we were doing the right thing by ACKing and sending an immediate BYE. However, we also were doing the wrong thing and queuing an answer frame, thus causing the call to be answered. This would cause the call to be hung up by the channel thread, thus resulting in a second BYE being sent out. In this fix, I also have set the hangupcause to be correct since the initial BYE being sent by Asterisk had an unknown hangup cause. I have changed to using "Bearer capabilty not available" since the call was hung up due to an SDP offer/answer error. (closes issue ASTERISK-22621) reported by Kinsey Moore ........ Merged revisions 400970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400971 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	23b6a34d45	chan_sip: Do not increment the SDP version between 183 and 200 responses. Bumping the SDP version number can cause interoperability problems since receivers of the responses will expect that a 200 SDP will be identical to a previous 183 SDP. (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches: dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418) ........ Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b8980c3be8	chan_sip: Don't ignore expires value in contact header if it lacks semicolon (closes issue ASTERISK-22574) Reported by: Filip Jenicek Patches: chan_sip_expires.patch uploaded by Filip Jenicek (license 6277) ........ Merged revisions 400469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	fb1c96baf4	chan_sip: Allow Asterisk to retry after 403 on register This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi ........ Merged revisions 400137 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	ee37433d8e	chan_sip: Increase some scratch buffer sizes dealing with caller id. * Eliminated an unnecessary initialization in check_user_full(). (closes issue ASTERISK-22477) Reported by: Michael Shepelev ........ Merged revisions 400013 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b9133abc09	chan_sip: Reject calls on 200 OKs if no SDP has been received When Asterisk receives a 200 OK in response to an invite, that peer should have sent an SDP at some point by then. If the channel has never received an SDP, media won't have been set and the remote address won't be known. Endpoints in general should not be doing this. This patch makes it so that Asterisk will simply hang up a call if it sends a 200 OK at this point. So far this odd behavior for endpoints has only been observed in tests which involved manually created SIP transactions in SIPp. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/ ........ Merged revisions 399939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	591c3b403e	chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok 1st Issue When a realtime peer sends an un-REGISTER request, Asterisk un-registers the peer but the database table record still has regseconds and fullcontact for the peer. This results in calls attempting to be routed to the peer which is no longer registered. The expected behavior is to get busy/congested when attempting to call an un-registered peer through the dialplan. What was discovered is that we are clearing out the peer's registration in the database in parse_register_contact() when calling expire_register() but then upon returning from parse_register_contact(), update_peer() is run which stores back in the database table regseconds and fullcontact. 2nd Issue The reporter pointed out that the 200 ok being returned by Asterisk after un-registering a peer contains a Contact header with ;expires= and the Expires header is not set to 0. This is actually a regression. Tests were created for this second issue (ASTERISK-22548). The tests have been reviewed and a Ship It! was received on those tests. This patch does the following: * Do not ignore the Expires header value even when it is set to 0. The patch sets the pvt->expiry earlier on in the function so that it is set properly and used. * If pvt->expiry is 0, do not call update_peer since that means the peer has already been un-registered and there is no need to update the database record again since nothing has changed. (closes issue ASTERISK-22428) Reported by: Ben Smithurst Tested by: Ben Smithurst, Michael L. Young Patches: asterisk-22428-rt-peer-update-and-expires-header.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2869/ ........ Merged revisions 399794 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	155e9f821c	chan_sip: Make direct media reinvites for T38 put Asterisk in the media path Prior to this patch, Asterisk would incorrectly use the previous endpoint addresses in SDP in spite of providing its own port. T38 is never meant to be done through directmedia and Asterisk should always be in the media path for these streams. (closes issue ASTERISK-17273) Reported by: Kevin Stewart (closes issue ASTERISK-18706) Reported by: Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/ ........ Merged revisions 399456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d91ceb38f5	chan_sip: Revert r398835 due to failing tests involving originate (issue ASTERISK-22424) Reported by: Jonathan Rose ........ Merged revisions 398977 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	0860ba2a1b	chan_sip: Reject calls without prior SDP on 200 OK If we receive a 200 OK without SDP, we will now check to see if the remote address has been established for that channel's RTP session and if the to tag for that channel has changed from the most recent to tag in a response less than 200. If either a change has been made since the last to-tag was received or the remote address is unset, then we will drop the call. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header ........ Merged revisions 398835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	71857a4a5e	Fix various memory leaks main/config.c - cleanup cache fie includes res/res_security_log.c - unregister logger level channesl/chan_sip.c - cleanup io context and notify_types main/translator.c - cleanup at shutdown main/named_acl.c - cleanup cli commands main/indications.c - ast_get_indication_tone() unref default_tone_zone if used (closes issues ASTERISK-22378) Reported by: Corey Farrell Patches: config_shutdown.patch uploaded by coreyfarrell (license 5909) res_security_log.patch uploaded by coreyfarrell (license 5909) chan_sip-11.patch uploaded by coreyfarrell (license 5909) indications_refleak.patch uploaded by coreyfarrell (license 5909) named_acl-cli_unreg-11.patch uploaded by coreyfarrell (license 5909) translate_shutdown.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 398102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@398103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c58bab8ce3	AST-2013-005: Fix crash caused by invalid SDP If the SIP channel driver processes an invalid SDP that defines media descriptions before connection information, it may attempt to reference the socket address information even though that information has not yet been set. This will cause a crash. This patch adds checks when handling the various media descriptions that ensures the media descriptions are handled only if we have connection information suitable for that media. Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing the solution to this problem. (closes issue ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches: issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397756 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397757 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	fdc86bb44c	Fix uninitialized value in struct ast_control_pvt_cause_code usage. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4fd979228d	AST-2013-004: Fix crash when handling ACK on dialog that has no channel A remote exploitable crash vulnerability exists in the SIP channel driver if an ACK with SDP is received after the channel has been terminated. The handling code incorrectly assumed that the channel would always be present. This patch adds a check such that the SDP will only be parsed and applied if Asterisk has a channel present that is associated with the dialog. Note that the patch being applied was modified only slightly from the patch provided by Walter Doekes of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches: issueA21064_fix.patch uploaded by wdoekes (License 5674) ........ Merged revisions 397710 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 397711 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	142c5d4816	Prevent a crash on outbound SIP MESSAGE requests. If a From header on an outbound out-of-call SIP MESSAGE were malformed, the result could crash Asterisk. In addition, if a From header on an incoming out-of-call SIP MESSAGE request were malformed, the message was happily accepted rather than being rejected up front. The incoming message path would not result in a crash, but the behavior was bad nonetheless. (closes issue ASTERISK-22185) reported by Zhang Lei git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	88a5f18dec	Fix Not Storing Current Incoming Recv Address In 1.8, r384779 introduced a regression by retrieving an old dialog and keeping the old recv address since recv was already set. This has caused a problem when a proxy is involved since responses to incoming requests from the proxy server, after an outbound call is established, are never sent to the correct recv address. In 11, r382322 introduced this regression. The fix is to revert that change and always store the recv address on incoming requests. Thank you Walter Doekes for helping to point out this error and Mark Michelson for your input/review of the fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin Tested by: Alex Zarubin, Karsten Wemheuer Patches: asterisk-22071-store-recvd-address.diff by Michael L. Young (license 5026) ........ Merged revisions 397204 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397205 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	3b91cde004	Remove REF_DEBUG definition. ........ Merged revisions 397156 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	e510fa1514	Fix refcounting of sip_pvt in test_sip_rtpqos test and unlink it from the list of pvts. (closes issue ASTERISK-22248) reported by Corey Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell (license #5909) ........ Merged revisions 397112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@397133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	f83b144899	chan_sip: Convert 'just did sched_add waitid...' from warning to debug message. Patches: reviewboard-2377.patch uploaded by Paul Belanger Review: https://reviewboard.asterisk.org/r/2377/ ........ Merged revisions 396582 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	16160ea357	chan_sip: Fix IP-addr in warning when rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/ ........ Merged revisions 396579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	1e03a50878	Fix Registration Failure When A Peer And TLS Are Used If a peer is used in a register line and TLS is defined as the transport, the registration fails since the transport on the dialog is never set properly resulting in UDP being used instead of TLS. This patch sets the dialog's transport based on the transport that was defined in the register line. If the register line does not specify a transport, the parsing function for the register line always defaults back to UDP. (closes issue ASTERISK-21964) Reported by: Doug Bailey Tested by: Doug Bailey Patches: asterisk-21964-set-reg-dialog-transport.diff by Michael L. Young (license 5026) ........ Merged revisions 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	eec46f56f4	Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix The prior code committed, r385473, failed to take into consideration that not all outgoing calls will be to a peer. My fault. This patch does the following: * Check if there is a related peer involved. If there is, check and set NAT settings according to the peer's settings. * Fix a problem with realtime peers. If the global setting has auto_force_rport set and we issued a "sip reload" while a peer is still registered, the peer's flags for NAT are reset to off. When this happens, we were always setting the contact address of the peer to that of the full contact info that we had. (closes issue ASTERISK-21374) Reported by: jmls Tested by: Michael L. Young Patches: asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2524/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	771ce9e1e7	Fix copy/paste error in one-touch-recording implementation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	527a611c80	chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out after retries fail RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription The problem is that the State Notify requests rely on the 200OK reponse for pacing control and to not confuse the notify susbsystem. The issue is, the pendinginvite isn't cleared if a response isn't received, thus further notify's are never sent. The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure. (closes issue ASTERISK-21677) Reported by: Dan Martens Tested by: Dan Martens, David Brillert, alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2475/ ........ Merged revisions 387875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	aec4d2f239	chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the interval when not the refresher RFC 4028 Section 10 if the side not performing refreshes does not receive a session refresh request before the session expiration, it SHOULD send a BYE to terminate the session, slightly before the session expiration. The minimum of 32 seconds and one third of the session interval is RECOMMENDED. Prior to this asterisk would refresh at 1/2 the Session-Expires interval, or if the remote device was the refresher, asterisk would timeout at interval end. Now, when not refresher, timeout as per RFC noted above. (closes issue ASTERISK-21742) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2488/ ........ Merged revisions 387344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	2846881045	chan_sip: Honor Session-Expires in 200OK response when it's a RE-INVITE when asterisk is the refresher. RFC 4028 Section 7.2 "UACs MUST be prepared to receive a Session-Expires header field in a response, even if none were present in the request." What changed After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher. Symptom: After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device may respond with a much lower Session-Expires (180 in our case) value that it is now using. Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE. After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response. Fix: handle_response_invite() when 200OK, remove check for outbound and reinvite. (closes issue ASTERISK-21664) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2463/ ........ Merged revisions 387312 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	95dcae4aa6	Prevent crash in 'sip show peers' when the number of peers on a system is large When you have lots of SIP peers (according to the issue reporter, around 3500), the 'sip show peers' CLI command or AMI action can crash due to a poorly placed string duplication that occurs on the stack. This patch refactors the command to not allocate the string on the stack, and handles the formatting of a single peer in a separate function call. (closes issue ASTERISK-21466) Reported by: Guillaume Knispel patches: fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387134 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	9d809c0f42	Fix Displaying Symmetric RTP Global Setting * Use comedia_string() to display correctly the symmetric rtp setting when running "sip show settings" git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	99f3a897fb	Change Case On Forcerport For Consistency * Change "ForcerPort" to "Forcerport" to match everywhere else it is displayed ........ Merged revisions 386483 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	f07cccecfd	Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX When we reload Asterisk or chan_sip, the flags force_rport and comedia that are turned on and off when using the auto_force_rport and auto_comedia nat settings go back to the default setting off. These flags are turned on when needed or off when not needed at the time that a peer registers, re-registers or initiates a call. This would apply even when only the default global setting "nat=auto_force_rport" is being used, which in this case would only affect the force_rport flag. Everything is good except for the following: The nat setting is set to auto_force_rport and auto_comedia. We reload Asterisk and the peer's registration has not expired. We load in the settings for the peer which turns force_rport and comedia back to off. Since the peer has not re-registered or placed a call yet, those flags remain off. We then initiate a call to the peer from the PBX. The force_rport and comedia flags stay off. If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not. This patch does the following: * Moves the checking of whether a peer is behind NAT into its own function * Create a function to set the peer's NAT flags if they are using the auto_* NAT settings * Adds calls in sip_request_call() to these new functions in order to setup the dialog according to the peer's settings (closes issue ASTERISK-21374) Reported by: Michael L. Young Tested by: Michael L. Young Patches: asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2421/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9511761e81	Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE When a BYE request is processed in chan_sip, the current SIP dialog is detached from its associated Asterisk channel structure. The tech_pvt pointer in the channel object is set to NULL, and the dialog persists for an RFC mandated period of time to handle re-transmits. While this process occurs, the channel is locked (which is good). Unfortunately, operations that are initiated externally have no way of knowing that the channel they've just obtained (which is still valid) and that they are attempting to lock is about to have its tech_pvt pointer removed. By the time they obtain the channel lock and call the channel technology callback, the tech_pvt is NULL. This patch adds a few checks to some channel callbacks that make sure the tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit callbacks, which would crash if AMI initiated a DTMF on the channel at the same time as a BYE was received from the UA. This patch also adds checks on sip_transfer (as AMI can also cause a callback into this function), as well as sip_indicate (as lots of things can queue an indication onto a channel). Review: https://reviewboard.asterisk.org/r/2434/ (closes issue ASTERISK-20225) Reported by: Jeff Hoppe ........ Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	74c57919a4	Fix For Not Overriding The Default Settings In chan_sip The initial report was that the "nat" setting in the [general] section was not having any effect in overriding the default setting. Upon confirming that this was happening and looking into what was causing this, it was discovered that other default settings would not be overriden as well. This patch works similar to what occurs in build_peer(). We create a temporary ast_flags structure and using a mask, we override the default settings with whatever is set in the [general] section. In the bug report, the reporter who helped to test this patch noted that the directmedia settings were being overriden properly as well as the nat settings. This issue is also present in Asterisk 1.8 and a separate patch will be applied to it. (issue ASTERISK-21225) Reported by: Alexandre Vezina Tested by: Alexandre Vezina, Michael L. Young Patches: asterisk-21225-handle-options-default-prob_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2385/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	ef79c00991	Address uninitialized conditional that valgrind found ........ Merged revisions 384162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b984d78c5c	AST-2013-003: Prevent username disclosure in SIP channel driver When authenticating a SIP request with alwaysauthreject enabled, allowguest disabled, and autocreatepeer disabled, Asterisk discloses whether a user exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The information is disclosed when: * A "407 Proxy Authentication Required" response is sent instead of a "401 Unauthorized" response * The presence or absence of additional tags occurs at the end of "403 Forbidden" (such as "(Bad Auth)") * A "401 Unauthorized" response is sent instead of "403 Forbidden" response after a retransmission * Retransmission are sent when a matching peer did not exist, but not when a matching peer did exist. This patch resolves these various vectors by ensuring that the responses sent in all scenarios is the same, regardless of the presence of a matching peer. This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of the testing and the solution to this problem was done by Walter as well - a huge thanks to his tireless efforts in finding all the ways in which this setting didn't work, providing automated tests, and working with Kinsey on getting this fixed. (closes issue ASTERISK-21013) Reported by: wdoekes Tested by: wdoekes, kmoore patches: AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674) AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1eff40f21d	Resolve deadlock between SIP registration and channel based functions In r373424, several reentrancy problems in chan_sip were addressed. As a result, the SIP channel driver is now properly locking the channel driver private information in certain operations that it wasn't previously. This exposed two latent problems either in register_verify or by functions called by register_verify. This includes: * Holding the private lock while calling sip_send_mwi_to_peer. This can create a new sip_pvt via sip_alloc, which will obtain the channel container lock. This is a locking inversion, as any channel related lock must be obtained prior to obtaining the SIP channel technology private lock. Note that this issue was already fixed in Asterisk 11. * Holding the private lock while calling sip_poke_peer. In the same vein as sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing the same locking inversion. Note that this locking inversion typically occured when CLI commands were run while a SIP REGISTER request was being processed, as many CLI commands (such as 'sip show channels', 'core show channels', etc.) have to obtain the channel container lock. (issue ASTERISK-21068) Reported by: Nicolas Bouliane (issue ASTERISK-20550) Reported by: David Brillert (issue ASTERISK-21314) Reported by: Badalian Vyacheslav (issue ASTERISK-21296) Reported by: Gabriel Birke ........ Merged revisions 383863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	4a50764715	tcptls: Prevent unsupported options from being set AMI, HTTP, and chan_sip all support TLS in some way, but none of them support all the options that Asterisk's TLS core is capable of interpreting. This prevents consumers of the TLS/SSL layer from setting TLS/SSL options that they do not support. This also gets tlsverifyclient closer to a working state by requesting the client certificate when tlsverifyclient is set. Currently, there is no consumer of main/tcptls.c in Asterisk that supports this feature and so it can not be properly tested. Review: https://reviewboard.asterisk.org/r/2370/ Reported-by: John Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........ Merged revisions 383165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	fb8760d679	When a session timer expires during a T.38 call, re-invite with correct SDP When a session timer expires during a dialog that has re-negotiated to T.38 and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP containing audio media only. This causes some hilarity with the poor fax session under weigh. This patch corrects that by sending T.38 parameters if we are in the middle of a T.38 session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches: dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418) ........ Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	77ca918044	Include the Username field in SIP Registry events when Status is registered In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed to include the Username field. Somehow, one of the events was missed. This patch corrects that - the Username field should be included in all AMI Registry events involving SIP registrations. (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by: Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) ........ Merged revisions 382847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	96c231fc18	chan_sip: Update the via header when relaying SMS MESSAGE Prior to this change, certain conditions for sending the message would result in an address of '(null)' being used in the via header of the SIP message because a NULl value of pvt->ourip was used when initially generating the via header. This is fixed by adding a call to build_via when the address is set before sending the message. (closes issue ASTERISK-21148) Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382739 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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