asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	2a77af9ed0	chan_rtp.c: Cleanup ast_request() parameter parsing. * Fixed NULL crash potential if parameters are missing. * Reordered some operations so further diagnostic messages can be more helpful. Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70	9 years ago
Richard Mudgett	ade5275a3e	parking.h: Update ast_parking_park_call() doxygen to reality. ASTERISK-26029 Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260	9 years ago
Alexei Gradinari	c378b00a83	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: I7091783a7150252de8eeb455115bd00514dfe843	9 years ago
Mark Michelson	1c02b19b79	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
Matt Jordan	e773e3a9bb	ARI: Add the ability to download the media associated with a stored recording This patch adds a new feature to ARI that allows a client to download the media associated with a stored recording. The new route is /recordings/stored/{name}/file, and transmits the underlying binary file using Asterisk's HTTP server's underlying file transfer facilities. Because this REST route returns non-JSON, a few small enhancements had to be made to the Python Swagger generation code, as well as the mustache templates that generate the ARI bindings. ASTERISK-26042 #close Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181	9 years ago
Joshua Colp	40cb032009	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	ff3cbc0046	Merge "res_pjsip_empty_info: Respond to empty SIP INFO packets"	9 years ago
Joshua Colp	e205eb55a4	Merge "res_pjsip: Endpoint IP Access Controls"	9 years ago
snuffy	9766a12b4c	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	acbaa1b0cf	Merge "udptl: Don't eat sequence numbers until OK is received"	9 years ago
Joshua Colp	5acb25722c	Merge "logger: Support JSON logging with Verbose messages"	9 years ago
Joshua Colp	b57032c364	Merge "res_hep: Provide an option to pick the UUID type"	9 years ago
Joshua Colp	1f36270b21	Merge "res/res_hep_pjsip: Fix reported local IP address when bound to 'any'"	9 years ago
Tzafrir Cohen	111c4b0324	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
Joshua Colp	d4b77dad1b	res_pjsip_exten_state: Use the extension for publishing to. This change uses the newly added multi-user support for outbound publish to publish to the specific user that an extension state change is for. This also extends the res_pjsip_outbound_publish support to include the user specific From and To URI information in the outbound publishing of extension state. Since the URI is used when constructing the body it is important to ensure that the correct local and remote URIs are used. Finally the max string growths for the dialog-info+xml body generator has been increased as through testing it has proven to be too conservative. ASTERISK-25965 Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1	9 years ago
Kevin Harwell	3905997bae	res_pjsip_outbound_publish: Add multi-user support per configuration Added a new multi_user option that when specified allows a particular configuration to be used for multiple users. It does this by replacing the user portion of the server uri with a dynamically created one. Two new API calls have been added in order to make use of the new functionality: ast_sip_publish_user_send - Sends an outgoing publish message based on the given user. If state for the user already exists it uses that, otherwise it dynamically creates new outbound publishing state for the user at that time. ast_sip_publish_user_remove - Removes all outbound publish state objects associated with the user. This essentially stops outbound publishing for the user. ASTERISK-25965 #close Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc	9 years ago
Joshua Colp	fc68291d13	Merge "CHANGES: Update formatting of items"	9 years ago
Joshua Colp	e2fc83af50	Merge "ARI: Add the ability to play multiple media URIs in a single operation"	9 years ago
Joshua Colp	4e7a1b4193	Merge "chan_sip: Prevent extra Session-Expires headers from being added"	9 years ago
George Joseph	6e5e84458f	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
Matt Jordan	52148d93f4	CHANGES: Update formatting of items * Provide consistent indenting of lines in bulleted paragraphs * Respect the 80 character column width * Group all like items together, e.g., all dialplan applications under "Applications", etc. * Use a single blank line to break up functionality changes within a larger section * Use two blanks lines to delineate larger sections Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647	9 years ago
Matt Jordan	03d88b5656	ARI: Add the ability to play multiple media URIs in a single operation Many ARI applications will want to play multiple media files in a row to a resource. The most common use case is when building long-ish IVR prompts made up of multiple, smaller sound files. Today, that requires building a small state machine, listening for each PlaybackFinished event, and triggering the next sound file to play. While not especially challenging, it is tedious work. Since requiring developers to write tedious code to do normal activities stinks, this patch adds the ability to play back a list of media files to a resource. Each of the 'play' operations on supported resources (channels and bridges) now accepts a comma delineated list of media URIs to play. A single Playback resource is created as a handle to the entire list. The operation of playing a list is identical to playing a single media URI, save that a new event, PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final media URI. When the entire list is finished being played, a PlaybackFinished event is raised. In order to help inform applications where they are in the list playback, the Playback resource now includes a new, optional attribute, 'next_media_uri', that contains the next URI in the list to be played. It's important to note the following: - If an offset is provided to the 'play' operations, it only applies to the first media URI, as it would be weird to skip n seconds forward in every media resource. - Operations that control the position of the media only affect the current media being played. For example, once a media resource in the list completes, a 'reverse' operation on a subsequent media resource will not start a previously completed media resource at the appropiate offset. - This patch does not add any new operations to control the list. Hopefully, user feedback and/or future patches would add that if people want it. ASTERISK-26022 #close Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f	9 years ago
George Joseph	5bd1bf2816	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
George Joseph	ae81b55361	res_pjsip_outbound_registration: Clean up state when registration is deleted Nothing was cleaning up the registration state object when ast_sorcery_delete was called on a registration. So, the registration was deleted from sorcery but the state object went right on refreshing the registration (or failing to refresh the registration) with the peer. * Added a 'deleted' observer on registration that removes the state object. ASTERISK-25964 #close Reported-by Matt Jordan Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23	9 years ago
zuul	040522100b	Merge "configs/samples/pjsip.conf.sample: Fix typo"	9 years ago
George Joseph	8b5cee4a4f	res_pjsip: Set TCP_NODELAY on TCP transports Although it's perfectly legal to place multiple SIP messages in the same packet, it can cause problems because the Linux default is to enable Path MTU Discovery which sets the Don't Fragment bit on the packets. If adding a second message to the packet causes the MTU to be exceeded, and the destination isn't equipped to send a FRAGMENTATION NEEDED response to a large packet, the packet will just be dropped. We can't specifically tell the stack to send only 1 message per packet, but we can turn on TCP_NODELAY when we create the transport. This will at least tell the stack to send packets as soon as possible. ASTERISK-26005 #close Reported-by: Ross Beer Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd	9 years ago
Matt Jordan	3522376512	logger: Support JSON logging with Verbose messages When `2d7a4a3357` was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96	9 years ago
Matt Jordan	a1803cb5f4	configs/samples/pjsip.conf.sample: Fix typo A ':' is not a valid token for starting a comment. Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad	9 years ago
Matt Jordan	d29c17834c	res/res_hep_pjsip: Fix reported local IP address when bound to 'any' When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its local address the 'any' address, as opposed to the IP address we actually received the packet on. This can cause some confusion in Homer, as it will dutifully report what we send it. This patch uses the PJSIP inspection routines to determine which IP address we probably received the packet on based on the remote party's IP address. In the event that this fails, it falls back to the IP address natively reported by the transport. Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3	9 years ago
Joshua Colp	acdd0ae993	Merge "logger: Add PID to syslog messages."	9 years ago
Sean Bright	14938184a3	res_ari: Correct Location headers returned by some ARI resources The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c	9 years ago
Matt Jordan	e06a23681c	res_hep: Provide an option to pick the UUID type At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c	9 years ago
zuul	d9c5882e69	Merge "config_transport: Tell pjproject to allow all SSL/TLS protocols"	9 years ago
Alexei Gradinari	69a85a519f	res_pjsip: Endpoint IP Access Controls With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02	9 years ago
zuul	7643dc44b2	Merge "pjsip_distributor: Add missing newline to NOTICE"	9 years ago
Joshua Colp	1bfe8602a5	Merge "basic-cfg: asterisk.conf: don't set languages"	9 years ago
Joshua Colp	da2506b2dc	Merge "basic-cfg: asterisk.conf: debug level 5 spams"	9 years ago
Joshua Colp	d733dccf81	Merge "basic-cfg: asterisk.conf: defaults of options"	9 years ago
zuul	17e38e7f92	Merge "followme: delete the right recorded name file"	9 years ago
zuul	c5cb9d120f	Merge "basic-cfg: asterisk.conf: remove [directories]"	9 years ago
Mark Michelson	fd3f70598d	Use doubles instead of floats for conversions when comparing strings. In 13.9.0, there was an issue where PJSIP contacts added to an AOR would be deleted at seemingly random times. One reason this was happening was because of an operation to retrieve the contacts whose expiration time was less than or equal to the current time. When retrieving existing contacts, the contact's expiration time and the current time were converted from a string to a float, and those two floats were compared. On some systems, including mine, this conversion was horribly off. For instance, I could regularly see the string "1463079214" get converted into 1463079168.000000. When switching from using a float to using a double, the conversion was as expected. Why was the conversion to float off? My best guess is that the conversion to float was attempting to store the entire value in the 23 bit significand of the IEEE-754 floating point number. In particular, if you take only the 23 most significant bits of 1463079214, you get the messed up 1463079168 that we were seeing in the conversion. It likely was possible to get a more precise value by composing the number using an exponent, but the conversion did not work that way. With a double, you have a 52 bit significand, allowing the entire value to fit there, and thereby allowing an accurate conversion. ASTERISK-26007 #close Reported by Greg Siemon Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070	9 years ago
zuul	dd354009d3	Merge "res_pjsip_outbound_registration: generate correct Contact URI for TLS"	9 years ago
George Joseph	4f8cfa0220	pjsip_distributor: Add missing newline to NOTICE There was a newline missing from the end of the "no matching endpoint" notice. Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181	9 years ago
Sebastian Damm	d14d1ba826	res_pjsip_outbound_registration: generate correct Contact URI for TLS There are two types of SIP URIs indicating a secure transport: * sips:user@example.org * sip:user@example.org;transport=tls When using a sips URI, Asterisk checks incoming INVITEs and answers from the other side for sips URIs, and rejects the packet if there are only sip URIs. So Asterisk should only generate a sips Contact URI if the other side supports it. This patch makes Asterisk generate either a sip or sips Contact URI depending on the format of the server URI. If you want a sip URI, use: server_uri=sip:example.org\;transport=tls If you want a sips URI, use: server_uri=sips:example.org ASTERISK-25990 #close Reported-by: Sebastian Damm Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2	9 years ago
Alexei Gradinari	9f996624b0	logger: Add PID to syslog messages. During refactoring of this support the addition of the PID to messages was removed. This change adds it back in. ASTERISK-25538 #close Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36	9 years ago
Matt Jordan	5236ffed97	configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER When running on a system that does not support or use AST_UNDEFINED_SANITIZER or AST_LEAK_SANITIZER, the configure script would incorrectly set those constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would cause menuselect to error out, complaining that a blank value is not a valid option. This patch corrects the issue by setting the value to 0 if the options that those constants enable/disable is not found. Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba	9 years ago
Joshua Colp	1be506d811	Merge "res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches"	9 years ago
Joshua Colp	086e311a75	Merge "res_pjsip_outbound_publishing: After unloading the library won't load again"	9 years ago
Joshua Colp	0863ccee09	Merge "res_pjsip_outbound_publish: Won't unload if condition wait times out"	9 years ago
Joshua Colp	09e22d7d6c	Merge "res_pjsip_outbound_publish: Ref leak in off nominal callback paths"	9 years ago

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