asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	1eba6abae5	Do not queue message requests that we do not respond to. If we receive a MESSAGE request that we cannot send a response to, we should not send the incoming MESSAGE to the dialplan. This commit should help the bouncing message_retrans test to pass consistently. ........ Merged revisions 434218 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	3439487a81	res_pjsip_messaging: Serialize outbound SIP MESSAGEs Outbound SIP MESSAGEs had the potential to be sent out of order from how they were specified in a set of dialplan steps. This change creates a serializer for sending outbound MESSAGE requests on. This ensures that the MESSAGEs are sent by Asterisk in the same order that they were sent from the dialplan. ASTERISK-24937 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4579 ........ Merged revisions 433968 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	0b62e41654	Add stateful PJSIP response API call, and use it for out-of-dialog responses. Asterisk had an issue where retransmissions of MESSAGE requests resulted in Asterisk processing the retransmission as if it were a new MESSAGE request. This patch fixes the issue by creating a transaction in PJSIP on the incoming request. This way, if a retransmission arrives, the PJSIP transaction layer will resend the response and Asterisk will not ever see the retransmission. ASTERISK-24920 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4532/ ........ Merged revisions 433619 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7e097bce86	Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks. Valgrind found some memory leaks associated with ast_pjsip_rdata_get_endpoint(). The leaks would manifest when sending responses to OPTIONS requests, processing MESSAGE requests, and res_pjsip supplements implementing the incoming_request callback. * Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in res/res_pjsip.c:supplement_on_rx_request(), res/res_pjsip/pjsip_options.c:send_options_response(), res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and res/res_pjsip_messaging.c:send_response(). * Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in res/res_pjsip_nat.c:nat_on_rx_message(). * Fixed inconsistent but benign return value in res/res_pjsip/pjsip_options.c:options_on_rx_request(). Review: https://reviewboard.asterisk.org/r/4511/ ........ Merged revisions 433222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	09bfe4b208	res_pjsip_refer: Fix crash from a REFER and BYE collision. Analyzing a one-off crash on a busy system showed that processing a REFER request had a NULL session channel pointer. The only way I can think of that could cause this is if an outgoing BYE transaction overlapped the incoming REFER transaction in a collision. Asterisk sends a BYE while the phone sends a REFER to complete an attended transfer. * Made check the session channel pointer before processing an incoming REFER request in res_pjsip_refer. * Fixed similar crash potential for res_pjsip supplement incoming request processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER messages. * Made res_pjsip_messaging respond to a message body too large with a 413 instead of ignoring it. ASTERISK-24700 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4417/ ........ Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	86a4ce4957	PJSIP: Enforce module load dependencies This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425691 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	47bf7efc4d	Multiple revisions 420089-420090,420097 ........ r420089 \| mjordan \| 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) \| 72 lines ARI: Add channel technology agnostic out of call text messaging This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy ........ r420090 \| mjordan \| 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) \| 2 lines Remove automerge properties :-( ........ r420097 \| mjordan \| 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) \| 2 lines test_message: Fix strict-aliasing compilation issue ........ Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	20750e261b	chan_sip.c: Fixed off-nominal message iterator ref count and alloc fail issues. * Fixed early exit in sip_msg_send() not destroying the message iterator. * Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy() tolerant of a NULL iter parameter in case ast_msg_var_iterator_init() fails. * Made ast_msg_var_iterator_destroy() clean up any current message data ref. * Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy() use iter instead of i. * Eliminated RAII_VAR usage in res_pjsip_messaging.c:vars_to_headers(). ........ Merged revisions 413139 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413142 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	2bf37a417d	Add a "message_context" option for PJSIP endpoints. ........ Merged revisions 411157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	7cbb6eab15	PJSIP: Add Path header support This adds Path support to chan_pjsip in res_pjsip_path.c with minimal additions in res_pjsip_registrar.c to store the path and additions in res_pjsip_outbound_registration.c to enable advertisement of path support to registrars and intervening proxies. Path information is stored on contacts and is enabled via Address of Record (AoRs) and Registration configuration sections. While adding path support, it became necessary to be able to add SIP supplements that handled messages outside of sessions, so a framework for handling these types of hooks was added in parallel to the already-existing session supplements and several senders of out-of-dialog requests were refactored as a result. (closes issue ASTERISK-21084) Review: https://reviewboard.asterisk.org/r/3050/ ........ Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	04c5c39d56	res_pjsip_messaging: potential for field values in from/to headers to be missing Added in ability to specify display name format ("name" <sip:name@ipaddr:port>) for a given URI and made sure it was fully propagated to the outgoing message. Also made it so outoing messages in res_pjsip always send as "sip:". (closes issue ASTERISK-22924) Reported by: Anthony Messina Review: https://reviewboard.asterisk.org/r/3094/ ........ Merged revisions 405266 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	661ac14911	documentation: Add PJSIP technology to messaging documentation ........ Merged revisions 403796 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	c602b086ed	res_pjsip_messaging: send message to a default outbound endpoint In some cases messages need to be sent to a direct URI (sip:<ip address>). This patch adds in that support by using a default outbound endpoint. When sending messages, if no endpoint can be found then the default one is used. To facilitate this a new default_outbound_endpoint option was added to the globals section for pjsip.conf. Review: https://reviewboard.asterisk.org/r/2944/ ........ Merged revisions 403680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	12a0edac69	pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to crash because they were trying to dereference a NULL pointer. In the case of res_pjsip_messaging it was attempting to "print" a contact header that did not exist. In fact contact headers should not be part of a SIP MESSAGE, so the offending code was simply removed. In the case of res_pjsip_header_funcs a null private channel tech was being passed to the function and then later dereferenced. Added null checks (and error logging) to the read/write function handlers to guard against crashing. (closes issue ASTERISK-22821) Reported by: Anthony Messina ........ Merged revisions 402757 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	4746c068dc	pjsip_messaging: Added debug for in dialog messaging (issue ASTERISK-22777) Reported by: Matt Jordan ........ Merged revisions 402265 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	0e1d364fbd	res_pjsip_messaging: Register message technology as pjsip pjsip's message technology was being registered as 'sip', which was causing it to not load due it conflicting with chan_sip's registered 'sip' technology for messaging. It now registers as 'pjsip'. However, due to this change the "to" field for outgoing pjsip messages need to be prefixed with 'pjsip:' instead of 'sip:'. Incoming messages to res_pjsip_messaging will automatically have their "to" fields altered in order to accommodate the change. Outgoing messages also handle changing it back to 'sip' before being sent so the pjsip library will properly handle it. (closes issue ASTERISK-22445) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2833/ ........ Merged revisions 399339 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	5b3441ae55	Fix crash in res_pjsip_outbound_registration when the remote server can not be resolved. This crash was caused by decrementing the reference count of a newly created message when it should not be. This change fixes that but also fixes all other cases where this was incorrectly done. (closes issue ASTERISK-22188) Reported by: Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_" are now "chan_pjsip_" * All files that were "res_sip" are now "res_pjsip" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_" are now "pjsip_" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

19 Commits (3f54af689f35faf691cde8ff1d92479c4bcf259e)