asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	6c4df2c704	channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario When we receive an SDP as part of an offer/answer for a peer/friend has been configured to require encryption, and that SDP offer/answer failed to provide acceptable crypto attributes, we currently issue a WARNING that uses the phrase "we" and "requested". In this case, both of those terms are ambiguous - the user will probably think "we" is Asterisk (it most likely isn't) and it may not be a "request", so much as an SDP that was received in some fashion. This patch makes the WARNING messages slightly less bad and a bit more accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	789d4379b6	channels/chan_sip: Fix crash when transmitting packet after thread shutdown When the monitor thread is stopped, its pthread ID is set to a specific value (AST_PTHREADT_STOP) so that later portions of the code can determine whether or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit failed to check for that value, checking instead only for AST_PTHREAD_STOP. Passing the invalid yet very specific value to pthread_kill causes a crash. This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that it doesn't attempt to poke the thread if the thread has already been stopped. ASTERISK-24800 #close Reported by: JoshE git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	e5d1dbafe0	channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB When a SIP device that has its registration stored in RealTime unregisters, the entry for that device is updated with blank values, i.e., "", indicating that it is no longer registered. Unfortunately, one of those values that is 'blanked' is the device's port. If the column type for the port is not a string datatype (the recommended type is integer), an ODBC or database error will be thrown. MariaDB does not coerce empty strings to a valid integer value. This patch updates the query run from chan_sip such that it replaces the port value with a value of '0', as opposed to a blank value. This is the value that other database backends coerce the empty string ("") to already, and the handling of reading a RealTime registration value from a backend already anticipates receiving a port of '0' from the backends. ASTERISK-24772 #close Reported by: Richard Miller patches: chan_sip.diff uploaded by Richard Miller (License 5685) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	bc1e13dfc3	channels/chan_sip: Ensure that a BYE is sent during INVITE w/Replaces transfer Consider a scenario where Alice and Bob have an established dialog with each other external to Asterisk. Bob decides to perform an attended transfer of Alice to Asterisk. In this case, Alice will send an INVITE with Replaces to Asterisk, where the Replaces specifies Bob's dialog with Asterisk. In this particular scenario, Asterisk will complete the transfer, but - since Bob's channel has had Alice masqueraded into it and is now a Zombie - a BYE request will not be sent. This patch fixes that issue by adding a new flag to chan_sip that tracks whether or not we have an INVITE with Replaces. If we do, the flag is used on the sip_pvt to ensure that a BYE request is sent, even if the channel has been masqueraded away. Review: https://reviewboard.asterisk.org/r/4362/ ASTERISK-22436 #close Reported by: Eelco Brolman Tested by: Jeremiah Gowdy, Kristian Høgh patches: asterisk-11-hangup-replaced-3.diff uploaded by Jeremiah Gowdy (License 6358) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Mark Michelson	47b83c9378	Use SIPS URIs in Contact headers when appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in chan_sip. ASTERISK-24646 #close Reported by Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/4346 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	9f694d888e	chan_sip: stale nonce causes failure When refreshing (with a small expiration) a registration that was sent to chan_sip the nonce would be considered stale and reject the registration. What was happening was that the initial registration's "dialog" still existed in the dialogs container and upon refresh the dialog match algorithm would choose that as the "dialog" instead of the newly created one. This occurred because the algorithm did not check to see if the from tag matched if authentication info was available after the 401. So, it ended up assuming the original "dialog" was a match and stopped the search. The old "dialog" of course had an old nonce, thus the stale nonce message. This fix attempts to leave the original functionality alone except in the case of a REGISTER. If a REGISTER is received if searches for an existing "dialog" matching only on the callid. If the expires value is low enough it will reuse dialog that is there, otherwise it will create a new one. ASTERISK-24715 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4367/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	08efda063a	Fix typo's (retrieve, specified, address). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	2fa4484340	chan_sip: Case insensitive comparison of "defaultuser" parameter. All the other configuration options are case insensitive, so this one should be too. ASTERISK-24355 #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded by HZMI8gkCvPpom0tM (License 6658) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5c7b830492	chan_sip: Send CANCEL via original INVITE destination even after UPDATE request Given the following scenario: * Three SIP phones (A, B, C), all communicating via a proxy with Asterisk * A call is established between A and B. B performs a SIP attended transfer of A to C. B sets the call on hold (A is hearing MOH) and dials the extension of C. While phone C is ringing, B transfers the call (that is, what we typically call a 'blond transfer'). * When the transfer completes, A hears the ringing of phone C, while B is idle. In the SIP messaging for the above scenario, a REFER request is sent to transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an UPDATE request to phone C to update party information. This update is sent directly to phone C, not through the intervening proxy. This has the unfortunate side effect of providing route information, which is then set on the sip_pvt structure for C. If someone (e.g. B) is trying to get the call back (through a directed pickup), Asterisk will send a CANCEL request to C. However, since we have now updated the route set, the CANCEL request will be sent directly to C and not through the proxy. The phone ignores this CANCEL according to RFC3261 (Section 9.1). This patch updates reqprep such that the route is not updated if an UPDATE request is being sent while the INVITE state is INV_PROCEEDING or INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent to the correct location. Review: https://reviewboard.asterisk.org/r/4279 ASTERISK-24628 #close Reported by: Karsten Wemheuer patches: issue.patch uploaded by Karsten Wemheuer (License 5930) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	822abf9e9b	Fix printf problems with high ascii characters after r413586 (1.8). In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) ptr); +out += sprintf(out, "%%%02X", (unsigned) ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	03d831ec94	chan_sip: Allow T.38 switch-over when SRTP is in use. Previously when SRTP was enabled on a channel it was not possible to switch to T.38 as no crypto attributes would be present. This change makes it so it is now possible. If a T.38 re-invite comes in SRTP is terminated since in practice you can't encrypt a UDPTL stream. Now... if we were doing T.38 over RTP (which does exist) then we'd have a chance but almost nobody does that so here we are. ASTERISK-24449 #close Reported by: Andreas Steinmetz patches: udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3aeed57b3c	res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero. Frames with a payload length of 0 were incorrectly handled in res_http_websocket. Provided a frame with a payload had been received prior it was possible for a double free to occur. The realloc operation would succeed (thus freeing the payload) but be treated as an error. When the session was then torn down the payload would be freed again causing a crash. The read function now takes this into account. This change also fixes assumptions made by users of res_http_websocket. There is no guarantee that a frame received from it will be NULL terminated. ASTERISK-24472 #close Reported by: Badalian Vyacheslav Review: https://reviewboard.asterisk.org/r/4220/ Review: https://reviewboard.asterisk.org/r/4219/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	094eeade6e	ast_str: Fix improper member access to struct ast_str members. Accessing members of struct ast_str outside of the string manipulation API routines is invalid since struct ast_str is supposed to be treated as opaque. Review: https://reviewboard.asterisk.org/r/4194/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	b3e0d05aad	chan_sip: Fix theoretical leak of p->refer. If transmit_refer is called when p->refer is already allocated, it leaks the previous allocation. Updated code to always free previous allocation during a new allocation. Also instead of checking if we have a previous allocation, always create a clean record. ASTERISK-15242 #close Reported by: David Woolley Review: https://reviewboard.asterisk.org/r/4160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cfa7763f85	channels/chan_sip: Add improved support for 4xx error codes This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER response handling. This helps interoperability in a number of scenarios. Review: https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch uploaded by oej (License 5267) ........ Merged revisions 426599 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bcd3f49994	channels/chan_sip: Support mutltiple Supported and Required headers A SIP request may contain multiple Supported: and Required: headers. Currently, chan_sip only parses the first Supported/Required header it finds. This patch adds support for multiple Supported/Required headers for INVITE requests. Review: https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close Reported by: Olle Johansson patches: rb2478.patch uploaded by oej (License 5267) ........ Merged revisions 426594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ebb3d8db82	channels/chan_sip: Respect outboundproxy setting when sending qualify requests The outboundproxy setting is currently ignored when sending OPTIONS requests as a result of the qualify setting. This means that if an Asterisk server is unable to send the packet directly to a peer, it is unable to qualify any non-inbound registered peer (e.g. a peer SIP Trunk). This patch grabs the outboundproxy information for a peer when a qualify attempt is being constructed and, if it finds the information, uses it when sending the OPTIONS request. Review: https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close Reported by: Damian Ivereigh patches: outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632) ........ Merged revisions 425818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	03a4a28a2e	chan_sip: Fix so asterisk won't send reINVITE after a BYE. After a reINVITE glare situation, Asterisk would re-send the reINVITE even though the call had been hung up in the mean time. This patch unschedules the reinvite when handling the BYE. ASTERISK-22791 #close Reported by: Paolo Compagnini Tested by: Paolo Compagnini Review: https://reviewboard.asterisk.org/r/4056/ (testcase is in review r4055) ........ Merged revisions 425296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	3b4fcfe6de	chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE. If a device re-INVITEs at the same time as the dialog is hung up, and if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would fail to destroy the dialog after a while. This resulted in (most prominently) file handle leaks. (Patch reindented by me.) ASTERISK-20784 #close ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334) patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418) Reviewboard: https://reviewboard.asterisk.org/r/4052/ (testcase can be found at r4051) ........ Merged revisions 425068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8b405b7b34	chan_sip: Clean leak on error path of process_sdp Resolve leak in process_sdp that occurs in 2 error path's where crypto lines are expected but not provided. ASTERISK-24385 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4045/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	2e62e7b1ef	chan_sip: Simplify some unref code by removing unlink_peer_from_tables. ASTERISK-22945 #related Reported by: ibercom Patches: asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599) ........ Merged revisions 424181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	a819ae6b8a	chan_sip: Remove excess ref of realtime peer before sip_poke_peer. The peer is referenced at the end of sip_poke_peer, it should not get an extra ref before the call to sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close Reported by: ibercom Tested by: Yuriy Gorlichenko Patches: asterisk11.patch uploaded by ibercom (License #6599) Review: https://reviewboard.asterisk.org/r/4031/ ........ Merged revisions 424176 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9bb255bc7c	chan_sip: Unref outbound proxy structure on dialog/pvt destruction. Make sure outbound proxy refs are always unreffed on dialog destruction. Review: https://reviewboard.asterisk.org/r/4016/ ........ Merged revisions 423800 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	559b43bd6d	chan_sip: On INVITE retransmission, don't add an extra 503 response. INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is retransmitted, asterisk would generate a 503 in addition to the 486. Thanks Torrey Searle for providing a working regression test. ASTERISK-24335 #close Review: https://reviewboard.asterisk.org/r/4003/ Patches: retrans_486_invite.patch uploaded by Torrey Searle (License #5334) ........ Merged revisions 423720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5cdd1413f1	CallerID: Fix parsing of malformed callerid This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	905f1f61e1	chan_sip: Use the server reflexive ICE candidate RTCP port as provided. This code originally worked around an issue within res_rtp_asterisk itself. The wrong socket was being used for the STUN check for RTCP, causing the port to be the same as RTP. This was subsequently fixed and the RTCP port provided for the ICE candidate is correct and does not need to be incremented. ASTERISK-23997 #close Reported by: Badalian Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav (license 5249) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421909 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6c4c543d48	chan_sip: Don't use port derived from fromdomain if it isn't set If a user does not provide a port in the fromdomain setting, chan_sip will set the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will then get used unilaterally in certain places. This causes issues with TLS, where the default port is expected to be 5061. This patch modifies chan_sip such that fromdomainport is only used if it is not the standard SIP port; otherwise, the port from the SIP pvt's recorded self IP address is used. Review: https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close Reported by: Elazar Broad patches: fromdomainport_fix.diff uploaded by Elazar Broad (License 5835) ........ Merged revisions 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	d779523c8c	chan_sip: Replace sip_tls_read() and resolve the large SDP poll issue. Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. ASTERISK-18345 #close Reported by: Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad Review: https://reviewboard.asterisk.org/r/3882/ ........ Merged revisions 420434 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@420435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	aaa44d6ea5	chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy sip_subscribe_mwi_destroy calls sip_destroy on the reference counted mwi->call. This results in the fields of mwi->call being freed, but mwi->call itself it leaked. If other code is still using mwi->call it can cause problems. This change uses dialog_unref instead, to balance the ref provided by sip_alloc(). ASTERISK-24087 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/ ........ Merged revisions 419440 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	915de454f8	res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1644cd874d	chan_sip: be more tolerant of whitespace between attributes in SDP fmtp line This patch is essentially a backport of a small portion of r397526 from ASTERISK-21981. In that patch, pass through support and format attribute negotiation was added for Opus. Part of that included being more tolerant to whitespace in the fmtp line of an SDP; that part of the patch is being applied here. As the author of the backport pointed out, in SDP, the fmtp line is allowed to include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001) includes an example for this. This was not removed in the updated RFC 4867 in 2007. Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 #close Reported by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520) ........ Merged revisions 417587 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	32d684ee25	res_http_websocket: Close websocket correctly and use careful fwrite When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	2875174e97	chan_sip: Fix handling of "From" headers longer than 256 characters From headers were processed using a 256 character buffer on the stack. This change replaces that with a heap allocation by ast_strdup. ASTERISK-23790 #close Reported by: uniken1 Tested by: uniken1 Review: https://reviewboard.asterisk.org/r/3669/ Patches: chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 417248 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@417249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	11553fd489	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ ........ Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@416067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	df686c50d8	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	a92d272d2f	chan_sip: Fix order of variables specified in SIPNotify action Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ ........ Merged revisions 415359 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	611f27fbd9	chan_sip: Start session timer at 200, not at INVITE. Asterisk started counting the session timer at INVITE while the other end correctly started at 200. This meant that for short session-expiries (90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk would wrongly assume that the timer was hit before the other end thought it was time to send a session refresh. This resulted in prematurely ended calls. This changes the session timer to start counting first at 200 like RFC says it should. (Also removed a few excess NULL checks that would never hit, because if they did, asterisk would have crashed already.) ASTERISK-22551 #close Reported by: i2045 Review: https://reviewboard.asterisk.org/r/3562/ ........ Merged revisions 414620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	93e4470a65	chan_sip: Add TLS and SRTP status to CLI command 'sip show channel' ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413876 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	adb50be36d	chan_sip+CEL: Add missing ANSWER and PICKUP events to INVITE/w/replaces pickup. When doing a "BLF-style call pickup" -- an INVITE with Replaces: header -- the CEL log would lack the ANSWER and PICKUP events. This patch adds the two missing events to the handle_invite_replaces() function. ASTERISK-22977 #close Review: https://reviewboard.asterisk.org/r/3073/ ........ Merged revisions 413832 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abac3330cf	Allow Asterisk to compile under GCC 4.10 This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	709d39b662	chan_sip.c: Fixed off-nominal message iterator ref count and alloc fail issues. * Fixed early exit in sip_msg_send() not destroying the message iterator. * Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy() tolerant of a NULL iter parameter in case ast_msg_var_iterator_init() fails. * Made ast_msg_var_iterator_destroy() clean up any current message data ref. * Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy() use iter instead of i. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	43f21b8564	chan_sip: Add sendrpid trust options In r411189, some behavior was changed which made sendrpid behavior act in a more trusting manner by sending full user data for peers set with private caller presence in P-Asserted-Identity headers. Since this changed long time expected behaviors, we decided to pull that patch when that was pointed out by the community. Instead, this patch provides a trust_id_outbound setting which will expose the data per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers at all if set to 'no'. By default trust_id_outbound will be set to 'legacy' which will preserve the behavior prior to these patches. Extra special thanks to Walter Doekes for providing advice and feedback. (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski Review: https://reviewboard.asterisk.org/r/3447/ ........ Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	bb3b4352ac	chan_sip.c: Moved some sip_pvt unrefs after their last use. * Moved sip_pvt unref in ast_hangup() and handle_request_do() to the end of the function. The unref needs to happen after the last use of the pointer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	1d1cc62fba	Reverting r411189 so that it can be put up for public review --- r411189 \| jrose \| 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) \| 12 lines chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior to this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) --- ........ Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	33d1220bee	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. Review: https://reviewboard.asterisk.org/r/3377/ ........ Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	45ebd29e10	internal_timing: Remove the option and always make it enabled if a timing module is loaded. The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8fe29356ac	Fix dialplan function NULL channel safety issues (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	6a4e040aa8	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) ........ Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	336ecce920	chan_sip: Fix incorrect use of timers If update_provisional_keepalive() is called while send_provisional_keepalive_full() is waiting on the PVT lock, then pvt->provisional_keepalive_sched_id will be changed to a new sched_id value by update_provisional_keepalive(), but that new sched_id then may be overwritten with -1 by send_provisional_keepalive_full(), killing the pvt's reference to a schedule and "leaking" the reference. (closes issue ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies Patches: provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012) ........ Merged revisions 411088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	4644a07334	chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted. (closes issue ASTERISK-20841) Reported by: Kelly Goedert ........ Merged revisions 411021 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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