asterisk

Commit Graph

Author	SHA1	Message	Date
Terry Wilson	b7233b18eb	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/ ........ Merged revisions 373652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373665 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373690 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	6f61cb50c5	Doxygen Updates - janitor work Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen. Further updates coming. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e8380afc8a	Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As mentioned on the review for this, WebRTC has moved towards choosing DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds support for this but makes it available for normal SIP clients as well. Testing has been done to ensure that this introduces no regressions with existing behavior and also that it functions as expected. Review: https://reviewboard.asterisk.org/r/2113/ ........ Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ae179ac5b4	Only re-create an SRTP session when needed 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. (issue ASTERISK-20194) Reported by: Nicolo Mazzon Tested by: Nicolo Mazzon Review: https://reviewboard.asterisk.org/r/2099 ........ Merged revisions 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372710 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372711 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8018b879a2	Clean up doxygen warnings This patch fixes numerous doxygen warnings across Asterisk. It also updates the makefile to regenerate the doxygen configuration on the local system before running doxygen to help prevent warnings/errors on the local system. Much thanks to Andrew for tackling one of the Asterisk janitor projects! (issue ASTERISK-20259) Reported by: Andrew Latham Patches: doxygen_partial.diff uploaded by Andrew Latham (license 5985) make_progdocs.diff uploaded by Andrew Latham (license 5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9ee8b3c0f6	Extend extension state callbacks to have more information. Quote from review board: This patch extends the extension state callbacks so that monitoring channels (as chan_sip) get more information of the devices which are responsible for an extension state change. The additional information is needed by chan_sip to present names/numbers of the caller and callee in an early-state SIP notification. Users of extenstion state callback not interested in the additional information are not affected by the changes. Motivation: to present the involved party's name/number in an early-state nofification (used by the notified device as a pickup offer) one after another so that a user can see which call he will pick up in an undirected pickup. Such a pickup offer to a user shall indicate the same call (number/name-A calls number/name-B) as the call which would be picked up when an undirected pickup is executed. Users interested in additional state info must use the new functions ast_extension_state_add_extended() resp. ast_extension_state_add_destroy_extended() to register an extended state callback. When the callback is registered this way, an extra member device_state_info of struct ast_state_cb_info is passed to the callback in addition to the aggregated extension state. This container holds an object for every device of the monitored extension hint consisting of the device name, the device state and a channel reference to the channel which (presumably) caused the device state. The information is used by chan_sip for early-state notifications. When the state of a device changes and the new state contains AST_EVENT_RINGING, an early-state notification is sent to the subscribed devices with the caller/callee names/numbers of the oldest ringing channel of the monitored extension. The notified user may then invoke a direct pickup, which will pickup exactly this channel. Users of the old non-extended callbacks will only be called when the aggregated state did change (same behavior as before). Users of the extended callback will also be called when the state is unchanged but does contain AST_EVENT_RINGING. That could be the case if two channels are ringing at one device and one of them hangs up, so the aggregated state does not change. This way the monitoring channel can create a new early-state notification with the now ringing party-ids. Review: https://reviewboard.asterisk.org/r/2048 This contribution comes from Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5c4578f4ad	Add named callgroups/pickupgroups This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9f0127f087	Multiple revisions 370769-370771 ........ r370769 \| mmichelson \| 2012-08-03 16:35:00 -0500 (Fri, 03 Aug 2012) \| 24 lines 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 ........ r370770 \| mmichelson \| 2012-08-03 16:39:35 -0500 (Fri, 03 Aug 2012) \| 3 lines Remove unused variable. ........ r370771 \| mmichelson \| 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) \| 5 lines Seriously? Another compilation error fixed. Somebody beat me. ........ Merged revisions 370769-370771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370772 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	4377d511ae	Add headers from SIPAddHeader to outbound REFER requests. This is a patch from kkm from review board. This is useful for adding headers to REFER requests that emanate from a Transfer() dialplan application call. This also fixes some uses of the Referred-by header, removing an extra set of angle brackets. I've modified the reporter's original patch to not require any additions to the sip_refer header and to just remove the referred_by_name from sip_refer since it is no longer needed or used. (closes Issue ASTERISK-17639) reported by Kirill Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff uploaded by Kirill Katsnelson (license #5845) Review: https://reviewboard.asterisk.org/r/1159 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	e5210366e4	Clean up chan_sip This clean up was broken out from https://reviewboard.asterisk.org/r/1976/ and addresses the following: - struct sip_refer converted to use the stringfields API. - sip_{refer\|notify}_allocate -> sip_{notify\|refer}_alloc to match other *alloc functions. - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not get_pidf_msg_text_body3 but get_content, to match add_content. - get_body doesn't get the request body, renamed to get_content_line. - get_body_by_line doesn't get the body line, and is just a simple if test. Moved code inline and removed function. - Remove camelCase in struct sip_peer peer state variables, onHold -> onhold, inUse -> inuse, inRinging -> ringing. - Remove camelCase in struct sip_request rlPart1 -> rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata to pvt->nonce because that is what it is, no need to update struct sip_pvt because _it already has a nonce field_. - Removed struct sip_pvt randdata stringfield. - Remove useless (and inconsistent) 'header' suffix on variables in handle_request_subscribe. - Use ast_strdupa on Event header in handle_request_subscribe to avoid overly complicated strncmp calls to find the event package. - Move get_destination check in handle_request_subscribe to avoid duplicate checking for packages that don't need it. - Move extension state callback management in handle_request_subscribe to avoid duplicate checking for packages that don't need it. - Remove duplicate append_date prototype. - Rename append_date -> add_date to match other add_xxx functions. - Added add_expires helper function, removed code that manually added expires header. - Remove _header suffix on add_diversion_header (no other header adding functions have this). - Don't pass req->debug to request handle_request_XXXXX handlers if req is also being passed. - Don't pass req->ignore to check_auth as req is already being passed. - Don't create a subscription in handle_request_subscribe if p->expiry == 0. - Don't walk of the back of referred_by_name when splitting string in get_refer_info - Remove duplicate check for no dialog in handle_incoming when sipmethod == SIP_REFER, handle_request_refer checks for that. Review: https://reviewboard.asterisk.org/r/1993/ Patch-by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	4d6b524b61	Prevent multiple local candidates from being added with the same information and add support for disabling ICE on a per-peer basis. (closes issue ASTERISK-20088) Reported by: wimpy Review: https://reviewboard.asterisk.org/r/2044/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e938737570	Add support for SIP over WebSocket. This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb. Review: https://reviewboard.asterisk.org/r/2008 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	10afdf3a2a	Named ACLs: Introduces a system for creating and sharing ACLs This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	d97e6c1401	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/ ........ Merged revisions 369557 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369558 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	7d9e0158c3	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 ........ Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369437 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	e0883154cf	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 ........ Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369353 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	bf6ef69702	Allow chan_sip to decline unwanted media streams This change replaces the static array of four representable media streams with an AST_LIST so that chan_sip can keep track of offered media streams. This allows chan_sip to deal with offers containing multiple same-type streams and many other situations without rejecting the SDP offer in its entirety, yet still generating a valid response. This also covers cases where Asterisk can not comprehend the offer if it is in the correct format. Previously, chan_sip would reject SDP offers or entirely ignore individual stream offers in an effort to be more compatible which would often result in invalid SDP responses. Review: https://reviewboard.asterisk.org/r/1988/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	166b4e2b30	Multiple revisions 369001-369002 ........ r369001 \| kpfleming \| 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) \| 11 lines 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. ........ r369002 \| kpfleming \| 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) \| 3 lines Add a script to enable finding source files without support-levels defined. ........ Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c6142cf2cc	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) ........ Merged revisions 368738 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368739 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	14a985560e	Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	463f9d729a	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. Review: https://reviewboard.asterisk.org/r/1954 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f454dceaf3	Re-add LastMsgsSent value for SIP peers Previously, MWI logic utilized a counter called 'lastmsgssent' to know whether or not MWI NOTIFY requests had been sent to a specific peer. When MWI notifications were changed to use the internal event framework, this value was no longer needed for its original purpose. Hence, it was no longer updated with the new/old message counts for a peer. The value was previously removed for Asterisk 10; however, since it was still present in Asterisk 1.8 and still useful for reporting purposes, it was decided to re-add the value. This patch re-adds the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show peer [peer]' command, and makes it so that the value of lastmsgssent is updated appropriately. The value should now display the new/old message counts for a particular peer. (closes issue ASTERISK-17866) Reported by: Steve Davies patches by: ast-17866-rb1272.patch (License #5041 by irroot) Modified slightly for this commit Review: https://reviewboard.asterisk.org/r/1939 ........ Merged revisions 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 367369 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367376 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7b51320642	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 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	6fc8e9928d	chan_sip: Fix a small TEST_FRAMEWORK related error that prevents compiling Introduced with r366842, a function call made only with TEST_FRAMEWORK enabled was missing an argument since the function arguments were changed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	54268bca4a	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. ........ Merged revisions 366882 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366884 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	cd37bec058	logger: Adds additional support for call id logging and chan_sip specific stuff This patch improves the handling of call id logging significantly with regard to transfers and adding APIs to better handle specific aspects of logging. Also, changes have been made to chan_sip in order to better handle the creation of callids and to enable the monitor thread to bind itself to a particular call id when a dialog is determined to be related to a callid. It then unbinds itself before returning to normal monitoring. review: https://reviewboard.asterisk.org/r/1886/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	dd81b047db	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) ........ Merged revisions 366167 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366168 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366169 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	781f4657b9	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) ........ Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	ae1502be33	Add support for lightweight NAT keepalive. If enabled using the keepalive option in sip.conf a small packet will be sent at a regular interval to keep the NAT mapping open. This is lightweight as the remote side does not need to parse and handle a SIP message. (closes issue AST-783) Review: https://reviewboard.asterisk.org/r/1756/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Michael L. Young	abf40d9b28	Add IPv6 address support to security events framework. The current Security Events Framework API only supports IPv4 when it comes to generating security events. This patch does the following: * Changes the Security Events Framework API to support IPV6 and updates the components that use this API. * Eliminates an error message that was being generated since the current implementation was treating an IPv6 socket address as if it was IPv4. * Some copyright dates were updated on files touched by this patch. (closes issue ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael L. Young Patches: security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	9ac6938e09	Missed lastinvite CSeq int to uint32_t change from Review: https://reviewboard.asterisk.org/r/1699/ ........ Merged revisions 359809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	1bef7695ce	Add a security event for the case where fake authentication challenge is sent. ........ Merged revisions 357318 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	299dd5d4fc	Adds an option to sip.conf that prevents diversion headers from being added. send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ebe2c33b72	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/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356690 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	670797e5da	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) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356605 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e5c51ee44c	Add auto_force_rport and auto_comedia NAT options This patch adds the auto_force_rport and auto_comedia NAT options. It also converts the nat= setting to a list of comma-separated combinable options: no, force_rport, comedia, auto_force_rport, and auto_comedia. nat=yes remains as an undocumented option equal to "force_rport,comedia". The first instance of 'yes' or 'no' in the list stops parsing and overrides any previously set options. If an auto_* option is specified with its non-auto_ counterpart, the auto setting takes precedence. This patch builds upon the patch posted to ASTERISK-17860 by JIRA user pedro-garcia. (closes issue ASTERISK-17860) Review: https://reviewboard.asterisk.org/r/1698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	3342183016	Add callbackextension matching & realtime callbackextensions This patch is based on the one by David Vossel, developer extrodinaire, at https://reviewboard.asterisk.org/r/344/. If multiple peers are defined with the same host/port, but differing callbackextensions, it chooses the peer with the matching callbackextension. Since callbackextension creates an outbound registration with the callbackextension as the Contact address, matching an incoming request by that (in addition to the host/port) makes a lot of sense. This patch also adds support for callbackextension to realtime by querying all peers with callbackextensions on reload and adding registrations for them. (closes issue ASTERISK-13456) Review: https://reviewboard.asterisk.org/r/344/ Review: https://reviewboard.asterisk.org/r/1717/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	0ccc1f5274	Merged revisions 353321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r353321 \| alecdavis \| 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) \| 25 lines Merged revisions 353320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r353320 \| alecdavis \| 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) \| 18 lines 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/trunk@353347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	e0ca02fe21	Merged revisions 352863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r352863 \| alecdavis \| 2012-01-27 13:08:03 +1300 (Fri, 27 Jan 2012) \| 19 lines Merged revisions 352862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r352862 \| alecdavis \| 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) \| 12 lines 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/trunk@352864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	cbe57b11cb	Fixes for sending SIP MESSAGE outside of calls. * Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA function in the authorization attempt. * Pass up better From header contents for SIP to use. Now is in the "display-name" <URI> format expected by MessageSend. (Note that this is a behavior change that could concievably affect some people.) * Block user from adding standard headers that are added automatically. (To, From,...) * Allow the user to override the Content-Type header contents sent by MessageSend. * Decrement Max-Forwards header if the user transferred it from an incoming message. * Expand SIP short header names so the dialplan and other code only has to deal with the full names. * Documents what SIP expects in the MessageSend(from) parameter. (closes issue ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/1683/ ........ Merged revisions 352520 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ae32acfa3e	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. ........ Merged revisions 351618 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 351646 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	9c161503dc	Set port to a default sane value if a bogus one is provided when parsing hostnames. ........ Merged revisions 350679 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350680 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	2b2d34b3c9	Constify tag argument in REF_DEBUG related code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	19a4928fee	INFO/Record request configurable to use dynamic features Adds two new options to SIP peers allowing them to specify features (dynamic or builtin) to use when sending INFO/record requests. Recordonfeature activates whatever feature is specified when recieving a record: on request while recordofffeature activates whatever feature is specified when receiving a record: off request. Both of these features can be disabled by setting the feature to an empty string. (closes issue ASTERISK-16507) Reported by: Jon Bright Review: https://reviewboard.asterisk.org/r/1634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	03596bcb47	chan_sip autocreatepeer=persist option for auto-created peers to survive reload This patch moves destruction of sip peers to immediately after the general section of sip.conf is read so that autocreatepeer setting can be read before deletion of peers. If autocreatepeer=persist at reload, then peers created by the autocreatepeer setting will be skipped when purging the current SIP peer list. (closes ASTERISK-16508) Reported by: Kirill Katsnelson Patches: 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill Katsnelson (license 5845) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	113612b9d6	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/ ........ Merged revisions 345273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345275 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	6ef49c3214	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) ........ Merged revisions 345160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345161 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	863d7189b9	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. ........ Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344839 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c225800646	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/ ........ Merged revisions 344385 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344386 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04f04ff39d	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/ ........ Merged revisions 344215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344216 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	e28d306387	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. ........ Merged revisions 343276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343277 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7f883ef495	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/ ........ Merged revisions 343220 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343221 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	10de040b6e	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. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b63c1cc545	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. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	c4a7d0e2c7	Merged revisions 338417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338417 \| irroot \| 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) \| 19 lines Merged revisions 338416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 \| irroot \| 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) \| 12 lines 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/trunk@338435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	5982bdcb7c	Merged revisions 337595,337597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 \| jrose \| 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) \| 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 \| jrose \| 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) \| 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	8493c46308	Merged revisions 336936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336936 \| irroot \| 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) \| 14 lines Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	404151ad65	New sip.conf option for setting default tonezone for channel or individual devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	ea5262eeb5	Merged revisions 332042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332042 \| mnicholson \| 2011-08-16 10:20:48 -0500 (Tue, 16 Aug 2011) \| 2 lines fix a code comment AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1858e274e3	Merged revisions 332027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332027 \| mnicholson \| 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) \| 9 lines Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 \| mnicholson \| 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) \| 2 lines 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/trunk@332028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	5c627eba2b	Remove lastmsgssent from sip it has not been working since 1.6	14 years ago
Richard Mudgett	14d510c5b7	Merged revisions 326291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 \| rmudgett \| 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) \| 23 lines 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/trunk@326321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	1d93d217f0	Merged revisions 325740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 \| kmoore \| 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) \| 7 lines 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/trunk@325741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	6693c49a6a	Merged revisions 324685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324685 \| dvossel \| 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) \| 8 lines 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/trunk@324689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	0bd877621e	Addition of "outofcall_message_context" sip.conf option. Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	3f4d0e8743	Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	0caae96609	Merged revisions 321273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321273 \| jrose \| 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) \| 3 lines 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/trunk@321289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Murawki	9a7f807278	Merged revisions 321155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321155 \| markm \| 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) \| 10 lines Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic. Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails. (closes issue #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f90bc95f0d	Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 \| jrose \| 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) \| 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	32d43ebe19	When a error in T.38 negotiation happens or its rejected on a channel the state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	9066db4329	Merged revisions 318720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318720 \| mnicholson \| 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) \| 4 lines 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/trunk@318785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f0f5e237bf	Merged revisions 317474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317474 \| russell \| 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) \| 2 lines Fix more "set but unused" warnings. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	e87639fc26	Merged revisions 315894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315894 \| mnicholson \| 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) \| 28 lines Merged revisions 315893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315893 \| mnicholson \| 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) \| 21 lines Merged revisions 315891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315891 \| mnicholson \| 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) \| 14 lines Fix our compliance with RFC 3261 section 18.2.2. This change optimizes the free_via() function and removes some redundant null checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using the port specified in the Via header for routing responses (even when maddr is not set). Also the htons() function is now used when setting the port. Additional documentation comments have been added in various places to make the logic in the code clearer. (closes issue #18951) Reported by: jmls Patches: issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	079e794b1c	Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 \| mnicholson \| 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) \| 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 \| mnicholson \| 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) \| 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 \| mnicholson \| 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) \| 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	b8b1d085db	Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	15f633294d	Merged revisions 311612 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311612 \| bbryant \| 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) \| 9 lines 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/trunk@311613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	82ef85f20b	Merged revisions 311558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r311558 \| twilson \| 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) \| 5 lines 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/trunk@311559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	f7b7223fb6	Merged revisions 310088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r310088 \| jrose \| 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) \| 9 lines 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/trunk@310089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	48a9694ed0	Merged revisions 304245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r304245 \| mnicholson \| 2011-01-26 14:43:27 -0600 (Wed, 26 Jan 2011) \| 20 lines Merged revisions 304244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304244 \| mnicholson \| 2011-01-26 14:42:16 -0600 (Wed, 26 Jan 2011) \| 13 lines Merged revisions 304241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 \| mnicholson \| 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) \| 6 lines This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header. ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	e706b5706e	According to section 19.1.2 of RFC 3261: For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	cc0b7e7df5	Some scheduler API cleanup and improvements. Previously, I had added the ast_sched_thread stuff that was a generic scheduler thread implementation. However, if you used it, it required using different functions for modifying scheduler contents. This patch reworks how this is done and just allows you to optionally start a thread on the original scheduler context structure that has always been there. This makes it trivial to switch to the generic scheduler thread implementation without having to touch any of the other code that adds or removes scheduler entries. In passing, I made some naming tweaks to add ast_ prefixes where they were not there before. Review: https://reviewboard.asterisk.org/r/1007/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	41b95ee887	Merged revisions 286931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 \| jpeeler \| 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) \| 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	1b2039e7db	Merged revisions 285006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285006 \| dvossel \| 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) \| 9 lines Disables auth_options_request option by default. The auth_options_request option was created to do authentication on OPTIONS request just like INVITES are done. Since it has been noted that some endpoints use OPTIONS requests as a way of qualifying a peer and that a 401 authentication response could result in interoperability issues, this option has been disabled by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	d17eded2e9	Merged revisions 284950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284950 \| dvossel \| 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) \| 14 lines authenticate OPTIONS requests just like we would an INVITE OPTIONS requests should be treated the same as an INVITE This includes authentication. This patch adds the ability for incoming out of dialog OPTION requests to be authenticated before providing a response indicating whether an extension is available or not. The authentication routine works the exact same way as it does for incoming INVITEs. This means that if a peer has 'insecure=invite' in their peer definition, the same will be true for the processing of the OPTIONS request. Review: https://reviewboard.asterisk.org/r/881/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	848135748f	Merged revisions 283559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r283559 \| dvossel \| 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) \| 16 lines Merged revisions 283558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r283558 \| dvossel \| 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) \| 10 lines Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	bcf5988caf	Merged revisions 283493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283493 \| dvossel \| 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) \| 2 lines Changes the default behavior for sip.conf's pedantic option from "no" to "yes". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
	a491cac965	Merged revisions 281687 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281687 \| simon.perreault \| 2010-08-11 09:30:59 -0400 (Wed, 11 Aug 2010) \| 9 lines Fix parsing of IPv6 address literals in outboundproxy (closes issue #17757) Reported by: oej Patches: 17757.diff uploaded by sperreault (license 252) sip.conf.diff uploaded by sperreault (license 252) Tested by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	1990c4347e	Merged revisions 281650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r281650 \| russell \| 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) \| 5 lines Change the default value for alwaysauthreject in sip.conf to "yes". (closes issue #17756) Reported by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@281651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	7a987a853e	Merged revisions 280269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280269 \| jpeeler \| 2010-07-28 15:49:26 -0500 (Wed, 28 Jul 2010) \| 2 lines Give test category missing leading slash ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4a98994542	Merged revisions 279568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279568 \| dvossel \| 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) \| 21 lines transaction matching using top most Via header This patch modifies the way chan_sip.c does transaction to dialog matching. Asterisk now stores information in the top most Via header of the initial incoming request and compares that against other Requests that have the same call-id. This results in Asterisk being able to detect a forked call in which it has received multiple legs of the fork. I completely stripped out the previous matching code and made the comparisons a little more explicit and easier to understand. My comments in the code should offer all the details involving this patch. This patch also fixes a bug with the usage of the OBJ-MULTIPLE flag to find multiple dialogs with the same call-id. Since the callback function was returning (CMP_MATCH \| CMP_STOP) only the first item found was being returned. I fixed this by making a new callback function for finding multiple dialogs that only returns (CMP_MATCH) on a match allowing for multiple items to be returned. Review: https://reviewboard.asterisk.org/r/776/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	dd9428666d	Merged revisions 279504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r279504 \| mmichelson \| 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) \| 14 lines Allow for systems without locale support to be usable. A recent change to SIP URI comparison code added a locale-specific string comparison to the mix, and certain systems do not support such functions. This fix allows for those systems to still use Asterisk 1.8 (closes issue #17697) Reported by: pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347) Tested by: mmichelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	d1ad460b3d	SIP URI comparison fixes. This initially was created to work around the issue of using a string comparison instead of a binary comparison for IP addresses. It evolved a bit when test cases were created and it was discovered that comparison of URI parameters was not working exactly as it should. sip_uri_cmp() and its helpers have been moved to reqresp_parser.c and a new test has been added. (closes issue #17662) Reported by: oej Review: https://reviewboard.asterisk.org/r/792 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	57a92a6a7c	Allow IPv6 addresses for UDPTL streams. Review: https://reviewboard.asterisk.org/r/795 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	cb5892bb67	Fix port setting of external address in SIP. There are two changes here: 1. Since the externip setting can now have a port attached to it, calling it "externip" is misleading. The option is now documented and parsed as "externaddr." This also extends to the "matchexterniplocally" setting. It is now documented and parsed as "matchexternaddrlocally." The old names for the options may still be used, but they are no longer used in the sip.conf.sample file. 2. If no port is set for the externaddr, and UDP is the transport to be used, then we will set the port of the externaddr to that of the udpbindaddr. This was how things worked prior to the IPv6 merge, so this is a regression fix. (closes issue #17665) Reported by: mmichelson Patches: 17665.diff#2 uploaded by pprindeville (license 347) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	2289649901	Fix up some weird indentation problems in reqresp_parser.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	e129b31fc6	Add ability to configure the Max-Forwards header in the dialplan, as well as in sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	23b6e621d2	chan_sip: RFC compliant retransmission timeout Retransmission of packets should not be based on how many packets were sent, but instead on a timeout period. Depending on whether or not the packet is for a INVITE or NON-INVITE transaction, the number of packets sent during the retransmission timeout period will be different, so timing out based on the number of packets sent is not accurate. This patch fixes this by removing the retransmit limit and only stopping retransmission after a timeout period is reached. By default this timeout period is 64*(Timer T1) for both INVITE and non-INVITE transactions. For more information on sip timer values refer to RFC3261 Appendix A. Review: https://reviewboard.asterisk.org/r/749/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	b42c6cab17	Revert early destruction of RTP sessions Some code improperly assumes that the sessions are still there, so revert the change until I can find all of them and fix them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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