asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	27bafc3a8b	res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. We must continue using the serializer that the original SUBSCRIBE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. The "sip_transaction Unable to register SUBSCRIBE transaction (key exists)" message is a notable symptom of this issue. Outgoing subscriptions still create the pjsip/pubsub/<endpoint> serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0	9 years ago
Richard Mudgett	16b08444da	pjsip_distributor.c: Consistently pick a serializer for messages. Incoming messages that are not part of a dialog or a recognized response to one of our requests need to be sent to a consistent serializer. Under load we may be queueing retransmissions before we can process the original message. We don't need to throw these messages onto random serializers and cause reentrancy and message sequencing problems. * Created a pool of pjsip/distributor serializers that get picked by hashing the call-id and remote tag strings of the received messages. * Made ast_sip_destroy_distributor() destroy items in the reverse order of creation. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I2ce769389fc060d9f379977f559026fbcb632407	9 years ago
Richard Mudgett	993b769524	pjsip_distributor.c: Ignore messages until fully booted. We should not be processing any incoming messages until we are fully booted. We may not have dialplan or other needed configuration loaded yet. ASTERISK-26089 #close Reported by: Scott Griepentrog ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264	9 years ago
zuul	ae8f6e996e	Merge "res_odbc: Implement a connection pool." into 13	9 years ago
Joshua Colp	321a9b128f	res_odbc: Implement a connection pool. Testing has shown that our usage of UnixODBC is problematic due to bugs within UnixODBC itself as well as the heavy weight cost of connecting and disconnecting database connections, even when pooling is enabled. For users of UnixODBC 2.3.1 and earlier crashes would occur due to insufficient protection of the disconnect operation. This was fixed in UnixODBC 2.3.2 and above. For users of UnixODBC 2.3.3 and higher a slow-down would occur under heavy database use due to repeated connection establishment. A regression is present where on each connection the database configuration is cached again, with the cache growing out of control. The connection pool implementation present in this change helps to mitigate these issues by reducing how much we connect and disconnect database connections. We also solve the issue of crashes under UnixODBC 2.3.1 by defaulting the maximum number of connections to 1, returning us to the previous working behavior. For users who may have a fixed version the maximum concurrent connection limit can be increased helping with performance. The connection pool works by keeping a list of active connections. If the connection limit has not been reached a new connection is established. If the connection limit has been reached then the request waits until a connection becomes available before continuing. ASTERISK-26074 #close ASTERISK-26054 #close Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff	9 years ago
Alexander Traud	c6ee4a0f44	res_srtp: Instead of libSRTP use OpenSSL as random source. Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore. Therefore, the symbol RAND_bytes is used instead of crypto_get_random. ASTERISK-24436 #close Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96	9 years ago
zuul	93237209eb	Merge "core/dial: New channel variable FORWARDERNAME" into 13	9 years ago
Joshua Colp	cda3385409	alembic: Fix migration. The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting to use UniqueConstraint and failing. It was not imported and after importing it also continued to fail. I've changed the script to use the explicit name of the constraint instead. Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9	9 years ago
Joshua Colp	4665aa89e3	Merge "pjsip_distributor.c: Use correct rdata info access method (Part 2)." into 13	9 years ago
zuul	49b1be8952	Merge "logging,cdr,cel: Fix stringfield memory leak." into 13	9 years ago
zuul	c4771ffb7c	Merge "pjproject_bundled: Move to pjproject 2.5" into 13	9 years ago
Richard Mudgett	e2132dd358	logging,cdr,cel: Fix stringfield memory leak. The stringfields refactor to allow adding stringfields to the end of a structure (`f6f4cf459f`) exposed some incomplete cleanup code by some stringfield users. The most noticeable leaker is the logging system where there is a leak for every log message generated. ASTERISK-26078 #close Reported by: Etienne Lessard Patches: jira_asterisk_26078_v13.patch (license #5621) patch uploaded by Richard Mudgett Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782	9 years ago
Alexei Gradinari	2de58c6d01	core/dial: New channel variable FORWARDERNAME Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2	9 years ago
Richard Mudgett	b2ce0e354b	pjsip_distributor.c: Use correct rdata info access method (Part 2). The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799	9 years ago
zuul	acb614d5ae	Merge "res_pjsip_mwi_body_generator: Re-order the body items" into 13	9 years ago
Joshua Colp	e3c9ad6382	Merge "res_pjsip: add "via_addr", "via_port", "call_id" to contact" into 13	9 years ago
zuul	eff382c72b	Merge "res_pjsip: Add clarifying documentation to PJSIP_HEADER help text" into 13	9 years ago
zuul	856d6e34c3	Merge "res_pjsip: chatty verbose messages" into 13	9 years ago
George Joseph	fe305ccf01	res_pjsip_mwi_body_generator: Re-order the body items Re-ordered the body items so Message-Account is second. Messages-Waiting: no Message-Account: sip:1571@<IP Removed>:5060 Voice-Message: 0/0 (0/0) ASTERISK-26065 #close Reported-by: Ross Beer Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3	9 years ago
George Joseph	e8abfdcdc5	pjproject_bundled: Move to pjproject 2.5 Although all the patches we had against 2.4.5 were applied by Teluu, a new bug was introduced preventing re-use of tcp and tls transports This patch removes all the previous patches against 2.4.5, updates the version to 2.5, and adds a new patch to correct the transport re-use problem. Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068	9 years ago
Rusty Newton	37d039fdf3	res_pjsip: Add clarifying documentation to PJSIP_HEADER help text Added notes about when you can read or write headers. Specifically about being able to read on the inbound channel and write on an outbound channel. ASTERISK-26063 #close Reported by: Private Name Tested by: Rusty Newton Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5	9 years ago
Joshua Colp	dc230f3b8f	Merge "app_voicemail: fix bugs, imap mm_status log change to debug" into 13	9 years ago
Richard Mudgett	03d5b3ce5c	pjsip_distributor.c: Use correct rdata info access method. The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2	9 years ago
Alexei Gradinari	859bbec09b	app_voicemail: fix bugs, imap mm_status log change to debug Fixed some bugs: - create dirpath when save downloading message from IMAP storage. - create IMAP folder if not exists when saving to IMAP storage - check if file successfully opened before write to it - some IMAP checks - remove non-standard flag 'Unseen' etc Change to debug IMAP mm_status log instead of verbose. Remove unused X-Asterisk-VM-Caller-channel message header for security reason. The clients should not know name of peer/endpoint. ASTERISK-26045 #close Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b	9 years ago
Alexei Gradinari	230686f4ec	res_pjsip: add "via_addr", "via_port", "call_id" to contact As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576	9 years ago
Alexei Gradinari	04c12561a7	res_pjsip: chatty verbose messages There are a lot of verbose messages about Endpoint and Contact status changes if there are many dynamic endpoints. The patch sets verbose level 2 for Endpoint status changes and verbose level 3 for Contact status changes. ASTERISK-26055 #close Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7	9 years ago
Corey Farrell	a42bea3314	threadpool: Fix potential data race. worker_start checked for ZOMBIE status without holding a lock. All other read/write of worker status are performed with a lock, so this check should do the same. ASTERISK-25777 #close Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781	9 years ago
zuul	87fe6b1622	Merge "func_odbc: single database connection should be optional" into 13	9 years ago
zuul	06c6478ad8	Merge "Makefile: remove OSARCH check for init install" into 13	9 years ago
zuul	373e68539b	Merge "func_curl: Don't trim response text on non-ASCII characters" into 13	9 years ago
zuul	80d1fc859e	Merge "parking.h: Update ast_parking_park_call() doxygen to reality." into 13	9 years ago
Tzafrir Cohen	a32616d60c	Makefile: remove OSARCH check for init install There are more specific checks for the platform. Specifically this allows installing OS/X init scripts. ASTERISK-26038 #close Change-Id: If08933621145b10362a0cfe73c079301d9c13f50 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	9 years ago
Ivan Poddubny	9ddaab789e	func_curl: Don't trim response text on non-ASCII characters The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of a signed comparison. ASTERISK-25669 #close Reported by: Jesper patches: strings.curl.trim.patch submitted by Jesper (License 5518) Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a	9 years ago
Richard Mudgett	9453d1187a	parking.h: Update ast_parking_park_call() doxygen to reality. ASTERISK-26029 Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260	9 years ago
Alexei Gradinari	cd89501d48	func_odbc: single database connection should be optional func_odbc was changed in Asterisk 13.9.0 to make func_odbc use a single database connection per DSN because of reported bug ASTERISK-25938 with MySQL/MariaDB LAST_INSERT_ID(). This is drawback in performance when func_odbc is used very often in dialplan. Single database connection should be optional. ASTERISK-26010 Change-Id: I57d990616c957dabf7597dea5d5c3148f459dfb6	9 years ago
Mark Michelson	c0b190dd9a	res_pjsip: Match dialogs on responses better. When receiving an incoming response to a dialog-starting INVITE, we were not matching the response to the INVITE dialog. Since we had not recorded the to-tag to the dialog structure, the PJSIP-provided method to find the dialog did not match. Most of the time, this was not a problem, because there is a fall-back that makes the response get routed to the same serializer that the request was sent on. However, in cases where an asynchronous DNS lookup occurs in the PJSIP core, the thread that sends the INVITE is not actually a threadpool serializer thread. This means we are unable to record a serializer to handle the incoming response. Now, imagine what happens when an INVITE is sent on a non-serialized thread, and an error response (such as a 486) arrives. The 486 ends up getting put on some random threadpool thread. Eventually, a hangup task gets queued on the INVITE dialog serializer. Since the 486 is being handled on a different thread, the hangup task can execute at the same time that the 486 is being handled. The hangup task assumes that it is the sole owner of the INVITE session and channel, so it ends up potentially freeing the channel and NULLing the session's channel pointer. The thread handling the 486 can crash as a result. This change has the incoming response match the INVITE transaction, and then get the dialog from that transaction. It's the same method we had been using for matching incoming CANCEL requests. By doing this, we get the INVITE dialog and can ensure that the 486 response ends up being handled by the same thread as the hangup, ensuring that the hangup runs after the 486 has been completely handled. ASTERISK-25941 #close Reported by Javier Riveros Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0	9 years ago
Joshua Colp	ddcf983e39	res_sorcery_astdb: Filter fields to only the registered ones. This change introduces the same filtering that is done in res_sorcery_realtime to the res_sorcery_astdb module. This allows persisted sorcery objects that may contain unknown fields to still be read in from the AstDB and used. This is particularly useful when switching between different versions of Asterisk that may have introduced additional fields. ASTERISK-26014 #close Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2	9 years ago
Joshua Colp	3296d2d194	Merge "res_pjsip_empty_info: Respond to empty SIP INFO packets" into 13	9 years ago
Joshua Colp	f09f923514	Merge "res_pjsip_outbound_publishing: After unloading the library won't load again" into 13	9 years ago
Joshua Colp	4e4a991d90	Merge "res_pjsip: Endpoint IP Access Controls" into 13	9 years ago
snuffy	39fedfa423	res_pjsip_empty_info: Respond to empty SIP INFO packets Some SBCs require responses to empty SIP INFO packets after establishing call via INVITE, if not responded to they may drop your call after unspecified timeout of X minutes. They are identified by having no Content-Type, check for this and respond with 200 - OK message. ASTERISK-24986 #close Reported-by: Ilya Trikoz, Federico Santulli Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0	9 years ago
Joshua Colp	7d986ff3f6	Merge "res_pjsip_outbound_publish: Ref leak in off nominal callback paths" into 13	9 years ago
Joshua Colp	1b7ba9bb00	Merge "udptl: Don't eat sequence numbers until OK is received" into 13	9 years ago
Joshua Colp	4a04a5a3ec	Merge "res/res_hep_pjsip: Fix reported local IP address when bound to 'any'" into 13	9 years ago
Joshua Colp	811a54836d	Merge "res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches" into 13	9 years ago
Joshua Colp	cceccd68ad	Merge "res_pjsip_outbound_publish: Potential crash due to off nominal path" into 13	9 years ago
Joshua Colp	4509aa890f	Merge "res_pjsip_outbound_publish: Won't unload if condition wait times out" into 13	9 years ago
Joshua Colp	480b8cda30	Merge "chan_sip: Prevent extra Session-Expires headers from being added" into 13	9 years ago
George Joseph	935e0496c4	udptl: Don't eat sequence numbers until OK is received Scenario: Local fax -> Asterisk w/ firewall -> Provider -> Remote fax * Local fax starts rtp call to remote fax * Remote fax starts t38 call back to local fax. * Local fax sends t38 no-signal to Asterisk before sending an OK. * udptl processes the frame and increments the expected sequence number. * chan_sip drops the frame because the call isn't up so nothing goes out the external interface to open the port for incoming packets. * Local fax sends OK and Asterisk sends OK to the remote fax. * Remote fax sends t38 packets which are dropped by the firewall. * Local fax re-sends t38 no-signal with the same sequence number. * udptl drops the frame because it thinks it's a dup. * Still no outgoing packets to open the firewall. * t38 negotiation fails. The patch drops frames t38 received before udptl sequence processing when the call hasn't been answered yet. The second no-signal frame is then seen as new and is relayed out the external interface which opens the port and allows negotiation to continue. ASTERISK-26034 #close Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9	9 years ago
George Joseph	77e8ec162b	chan_sip: Prevent extra Session-Expires headers from being added When chan_sip does a re-INVITE to refresh a session and authentication is required, the INVITE with the Authorization header containes a second Session-Expires header without the ";refersher=" parameter. This is causing some proxies to return a 400. Also, when Asterisk is the uas and the refresher, it is including the Session-Expires and Min-SE headers in OPTIONS messages which is not allowed per RFC4028. This patch (based on the reporter's) Checks to see if a Session-Expires header is already in the message before adding another one. It also checks that the method is INVITE or UPDATE. ASTERISK-26030 #close Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9	9 years ago

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