103 Commits (ba65f21e73e9a8769baf6cf4b1e08e06e68862a5)

Author	SHA1	Message	Date
Matthew Jordan	a73c5b80ff	Resolve crashes due to large stack allocations when using TCP ... Asterisk had several places where messages received over various network transports may be copied in a single stack allocation. In the case of TCP, since multiple packets in a stream may be concatenated together, this can lead to large allocations that overflow the stack. This patch modifies those portions of Asterisk using TCP to either favor heap allocations or use an upper bound to ensure that the stack will not overflow: * For SIP, the allocation now has an upper limit * For HTTP, the allocation is now a heap allocation instead of a stack allocation * For XMPP (in res_jabber), the allocation has been eliminated since it was unnecesary. Note that the HTTP portion of this issue was independently found by Brandon Edwards of Exodus Intelligence. (issue ASTERISK-20658) Reported by: wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches: ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	33e5e3a4f3	Add "Require: timer" to 200 OK responses when appropriate. ... The method by which the Require header is added to 200 responses is inspired by the method that Olle Johansson uses in his darjeeling-prack branch. (closes issue ASTERISK-20570) Reported by Matt Jordan, at the behest of Olle Johansson Review: https://reviewboard.asterisk.org/r/2172 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	6f943721f1	Properly handle UAC/UAS roles for SIP session timers ... The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder. This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties. (closes issue AST-922) Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/2118/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	95f4884fe8	Only re-create an SRTP session when needed; respond with correct crypto policy ... In r356604, SRTP handling was fixed to accomodate multiple crypto keys in an SDP offer and the ability to re-create an SRTP session when the crypto keys changed. In certain circumstances - most notably when a phone is put on hold after having been bridged for a significant amount of time - the act of re-creating the SRTP session causes problems for certain models of phones. The patch committed in r356604 always re-created the SRTP session regardless of whether or not the cryptographic keys changed. Since this is technically not necessary, this patch modifies the behavior to only re-create the SRTP session if Asterisk detects that the remote key has changed. This allows models of phones that do not handle the SRTP session changing to continue to work, while also providing the behavior needed for those phones that do re-negotiate cryptographic keys. In addition, in Asterisk 1.8 only, it was found that phones that offer AES_CM_128_HMAC_SHA1_32 will end up with no audio if the phone is the initiator of the call. The phone will send an INVITE request specifying that AES_CM_128_HMAC_SHA1_32 be used for the cryptographic policy; Asterisk will set its policy to that value. Unfortunately, when the call is Answered and a 200 OK is sent back to the UA, the policy sent in the response's SDP will be the hard coded value AES_CM_128_HMAC_SHA1_80. This potentially results in Asterisk using the INVITE request's policy of AES_CM_128_HMAC_SHA1_32, while the phone uses Asterisk's response of AES_CM_128_HMAC_SHA1_80. Hilarity ensues as both endpoints think the other is crazy. This patch fixes that by caching the policy from the request and responding with it. Note that this is not a problem in Asterisk 10 and later, as the ability to configure the policy was added in that version. (issue ASTERISK-20194) Reported by: Nicolo Mazzon Tested by: Nicolo Mazzon Review: https://reviewboard.asterisk.org/r/2099 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9d402e02a5	Seriously? Another compilation error fixed. ... Somebody beat me. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	1119a574ed	Fix error in the "IPorHost" section of a SIP dialstring. ... This is based on the review request posted by Walter Doekes (referenced lower in the commit message) The main fix here is to treat the IPorHost portion of the dial string as a temporary outbound proxy. This ensures requests get sent to the proper location. Due to the age of the request, some parts were no longer relevant. For instance, the request moved outbound proxy parsing code into a single method. This is done in a previous commit, so it was not necessary to do again. Also, the review request fixed some errors with regards to request routing for CANCEL and ACK requests. This has also been fixed in more recent commits. (closes issue ASTERISK-19677) reported by Walter Doekes Review https://reviewboard.asterisk.org/r/1859 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	80efa31733	Help mitigate potential reinvite glare scenarios. ... When Asterisk servers are set up back-to-back, and direct media is to be used betweeen endpoints, it is fairly common for the two Asterisk servers to send direct media reinvites to each other simultaneously. This results in 491s and ACKs being exchanged between the servers. While the media eventually gets set up properly, the problem is that there can be a noticeable delay for the streams to stabilize. This patch adds a new directmedia option called "outgoing". With this set, an immediate direct media reinvite will only be sent if the call direction is outgoing. For incoming dialogs, an immediate direct media reinvite will not be sent, but further "reactionary" direct media reinvites may be sent. For those who are having some deja vu, that's because this patch was originally committed to trunk since there is a new configuration option added. After seeing a bug report about audio being slow to set up on SIP calls, it became apparent that this patch would be the best solution for resolving the issue. The patch is unintrusive and will have no effect unless the option is explicitly enabled. (closes issue AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	dff0057eba	Better handle re-INVITEs with provisional but no final repsonses ... A previous attempt at fixing this issue had negative side effects related to attended transfers which this patch should resolve. Many thanks to Steve Davies for all of the good suggestions and testing. (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson Review: https://reviewboard.asterisk.org/r/2009/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	16a61aecde	AST-2012-010: Clean up after a reinvite that never gets a final response ... The basic problem is that if a re-INVITE is sent by Asterisk and it receives a provisional response, but no final response, then the dialog is never torn down. In addition to leaking memory, this also leaks file descriptors and will eventually lead to Asterisk no longer being able to process calls. This patch just keeps track of whether there is an outstanding re-INVITE, and if there is goes ahead and cleans up everything as though there was no outstanding reinvite. Review: https://reviewboard.asterisk.org/r/2009/ (closes issue ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies, Terry Wilson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9b9a84740b	Re-fix how local tag is generated when sending a 481 to an INVITE. ... Match our local tag to whatever to-tag was sent in the initial INVITE. Because the size of the to-tag may not fit in the buffer in the sip_pvt, it has been changed to a string field. (closes issue ASTERISK-19892) reported by Walter Doekes Review: https://reviewboard.asterisk.org/r/1977 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	f83d1b98e8	Add support-level indications to many more source files. ... Since we now have tools that scan through the source tree looking for files with specific support levels, we need to ensure that every file that is a component of a 'core' or 'extended' module (or the main Asterisk binary) is explicitly marked with its support level. This patch adds support-level indications to many more source files in tree, but avoids adding them to third-party libraries that are included in the tree and to source files that don't end up involved in Asterisk itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	0353a57671	Fix coverity UNUSED_VALUE findings in core support level files ... Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	3bdb733c5d	Reorder and renumber tests appropriately ... It appears that a patch did not apply properly when adding tests 12 and 13 and test 11 was duplicated. These tests have been reordered and renumbered such that they make sense. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3edf245601	Fix a variety of memory leaks ... This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	a94fcae21b	Resolve FORWARD_NULL static analysis warnings ... This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	83d3444284	Fix many issues from the NULL_RETURNS Coverity report ... Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	1ded276c97	Missed lastinvite CSeq int to uint32_t change ... from Review: https://reviewboard.asterisk.org/r/1699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	534213a074	Fix worker thread resource leak in SIP TCP/TLS. ... The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	032962f1a2	Allow SRTP policies to be reloaded ... Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	8fc0050b54	RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer ... * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	4d2f8a9cfd	rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. ... If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f3c3de4c71	Misc minor fixes in reqresp_parser.c and chan_sip.c. ... * Fix corner cases in get_calleridname() parsing and ensure that the output buffer is nul terminated. * Make get_calleridname() truncate the name it parses if the given buffer is too small rather than abandoning the parse and not returning anything for the name. Adjusted get_calleridname_test() unit test to handle the truncation change. * Fix get_in_brackets_test() unit test to check the results of get_in_brackets() correctly. * Fix parse_name_andor_addr() to not return the address of a local buffer. This function is currently not used. * Fix potential NULL pointer dereference in sip_sendtext(). * No need to memset(calleridname) in check_user_full() or tmp_name in get_name_and_number() because get_calleridname() ensures that it is nul terminated. * Reply with an accurate response if get_msg_text() fails in receive_message(). This is academic in v1.8 because get_msg_text() can never fail. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	f7876c1dfe	Set port to a default sane value if a bogus one is provided when parsing hostnames. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7c0ead6cb8	Fixed crash from orphaned MWI subscriptions in chan_sip ... This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. When a peer is removed, either by pruning realtime SIP peers or by unloading / loading chan_sip, the MWI subscriptions that were orphaned would still be on the event engine list of valid subscriptions but have a pointer to a peer that no longer was valid. When an MWI event would occur, this would cause a seg fault. (closes issue ASTERISK-18663) Reported by: Ross Beer Tested by: Ross Beer, Matt Jordan Patches: blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) Review: https://reviewboard.asterisk.org/r/1610/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f435f45c50	Restore SIP DTMF overlap dialing method. ... The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support working correctly removed a long standing ability to do overlap dialing using DTMF in the early media phase of a call. See ASTERISK-18702 it has a very good description of the issue. I started with Pavel Troller's chan_sip.diff patch on issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf allowoverlap config option. The new option value causes the Incomplte application to not send anything with chan_sip so the caller can supply more digits via DTMF. * Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE since that is what it really means. * Fixed get_destination() inconsistency with the pickup extension matching. * Fixed initialization of PAGE3 of global_flags in reload_config(). (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: https://reviewboard.asterisk.org/r/1517/ Review: https://reviewboard.asterisk.org/r/1582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	f1762a4633	Update reqresp_parser parse_uri doxygen comments. ... The issue mentioned in the bug report had been fixed recently by twilson. The reporter included this documentation fix. (closes issue ASTERISK-18572) Reported by: Richard Miller Patch by: Richard Miller (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	0b3270601a	Remove unneeded if(params) checks in reqresp_parser. ... Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent reason. There is no way that params could become NULL in that piece of code, so I removed these excess checks again. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	47cff21b6d	Fix several bugs with SDP parsing and well-formedness of responses ... Fix bug ASTERISK-16558 which dealt with the order of responses to incoming streams defined by SDP. Fix unreported bug where offering multiple same-type streams would cause Asterisk to reply with an incorrect SDP response missing one or more streams without a proper declination. Fix bugs related to a single non-audio stream being offered with responses requesting codecs that were not offered in the initial invite along with an additional audio stream that was not in the initial invite. Review: https://reviewboard.asterisk.org/r/1516/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	a2a11ee737	Don't treat a host:port string as a domain ... The domain matching code prior to 1.8 used to manually remove the port from the host:port string when determining if an incoming request matched the list of domains. When switching to the new parsing functions, the documentation implied that the "domain" was being returned by these functions, when instead it was returning the "hostport" as defined by RFC 3261. This led to confusion and resulted in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set in sip.conf. This patch renames the "domain" variables in the parsing functions to "hostport" to more accurately describe what it is that they are returning and also properly truncates the resulting hostport strings when dealing with domain matching. Review: https://reviewboard.asterisk.org/r/1574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	702726c917	Make room for the fax detect flags ... The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5844133aba	Remove registertrying option in chan_sip ... This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f2b371fedf	More parking issues. ... * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a458ac621e	Fix some potential deadlocks pointed out by helgrind. ... * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e7d6d7ee19	The rtptimeout setting is ignored on a per peer basis. ... Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	c9325708c8	default 'sipstorecause' to no ... We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html (issue AST-580) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	3d709a2b55	use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option ... AST-580 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	48e78804e2	Used auth= parameter freed during "sip reload" causes crash. ... If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	484a8a8363	chan_sip: cleanup from the introduction of ast_str ... Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	5a8af0d613	Fixes sip crash when calling remove_uri_parameters with NULL ... AST-2011-009 (closes issue ASTERISK-18017) Reported by: jaredmauch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	a99d9c7770	markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Murawki	a35ebe0f61	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/branches/1.8@321155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	b3a2f27111	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/branches/1.8@319938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1b1961f73f	Handle ipv6 addresses in the sent-by Via: field. ... This change fixes a regression in via header parsing and ipv6 handling. (closes issue #18951) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	1ccfa50ba8	Fix more "set but unused" warnings. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	e8210addf8	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/branches/1.8@315894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	4468fe047e	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/branches/1.8@314628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Brett Bryant	a54ab29087	Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null ... value. (closes issue #18821) Reported by: cmaj Patches: patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2f95620a2f	Don't use static declared buf in parse_name_andor_addr ... This function isn't used anywhere yet, but we definitely don't want to keep the same value for buf between calls to the function. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ed3e04e831	Returns with an error notice if CHANNEL function of SIP channel is read without arguments. ... (Closes issue #18653) Reported by: wuwu Patches: diff.patch uploaded by jrose (license 1225) Tested by: jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	15b9d1ac10	Merged revisions 304244 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago