29173 Commits (8cb4f9cea1f8a34d2383551a971a86d3be02c4fd)

 

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Author	SHA1	Message	Date
Richard Mudgett	8cb4f9cea1	CHANNEL(callid): Give dialplan access to the callid. ... * Added CHANNEL(callid) to retrieve the call identifier log tag associated with the channel. Dialplan now has access to the call log search key associated with the channel so it can be saved in case there is a problem with the call. ASTERISK-26878 Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f	8 years ago
Joshua Colp	732367e806	Merge "res_pjsip: Symmetric transports"	8 years ago
Joshua Colp	76e64f5589	Merge "RFC sdp: Initial SDP creation"	8 years ago
George Joseph	5013d8f5d3	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	8 years ago
Joshua Colp	84f0871cba	Merge "Add rtcp-mux support"	8 years ago
Joshua Colp	aec2a087ed	Merge "chan_iax2: Reload of iax peer results in loss of host address/port"	8 years ago
zuul	5fdd61b556	Merge "res/res_pjsip_refer: call xfer w/o extension"	8 years ago
zuul	300b257045	Merge "app_queue: Handle the caller being redirected out of a queue bridge"	8 years ago
zuul	bcc9a07db2	Merge "funcs/func_devstate: Remove new line in Device field of during module load"	8 years ago
zuul	3f30ce1272	Merge "pbx.c: Fix crash from malformed exten pattern."	8 years ago
zuul	941671b27a	Merge "res_pjsip_endpoint_identifier_ip: Don't output error if no header_match."	8 years ago
zuul	3fe1d8afba	Merge "core: Add stream topology changing primitives with tests."	8 years ago
Mark Michelson	10fa49e327	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	8 years ago
Joshua Colp	e536ef7afb	Merge "res_pjsip_endpoint_identifier_ip: Add an option to match requests by header"	8 years ago
Joshua Colp	2cb449a621	Merge "configure: Don't use the progress bar with curl when downloading to stdout"	8 years ago
Torrey Searle	dc4cdafd42	res/res_pjsip_refer: call xfer w/o extension ... When transfering to a URI without an extension, ensure that the s extension of the dialplan is entered ASTERISK-26869 #close Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525	8 years ago
Sean Bright	982d6173c5	app_queue: Handle the caller being redirected out of a queue bridge ... A caller can leave the Queue() application after being bridged with a member in a few ways: * Caller or member hangup * Caller is transferred somewhere else (blind or atx) * Caller is externally redirected elsewhere The first 2 scenarios are currently handled by subscribing to stasis messages, but the 3rd is not explicitly covered. If a caller is redirected away from the Queue() application, the member who was last bridged with that caller will remain in an "In use" state until the caller hangs up. This patch adds handling of the caller leaving the queue via redirection. We monitor the caller-member bridge, and if the caller is the one that leaves, we treat it the same as we would a caller hangup. ASTERISK-26400 #close Reported by: Etienne Lessard Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334	8 years ago
Joshua Colp	0b8a57af6d	res_pjsip_endpoint_identifier_ip: Don't output error if no header_match. ... This change ensures that if no header_match option is set on an identify an error message is not output stating the option is set to an invalid value. ASTERISK-26863 Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a	8 years ago
Matt Jordan	1475604eff	res_pjsip_endpoint_identifier_ip: Add an option to match requests by header ... This patch adds a new features to the endpoint identifier module, 'match_header'. When set, inbound requests are matched by a provided SIP header: value pair. This option works in conjunction with the existing 'match' configuration option, such that if any 'match*' attribute matches an inbound request, the request is associated with the specified endpoint. Since this module now identifies by more than just IP address, appropriate renaming of the module and/or variables can be done in a non-release branch. ASTERISK-26863 #close Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 (cherry picked from commit 30f52d79d7)	8 years ago
George Joseph	71cc3fd969	Merge "res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue"	8 years ago
Joshua Colp	c5e1e85355	Merge "configs/samples/hep.conf.sample: Clarify how the HEP stack works"	8 years ago
Joshua Colp	2ffce60844	Merge "main/stasis_cache: Demote the ERROR message when removing a nonexistent item"	8 years ago
zuul	c152329932	Merge "res_pjsip_transport_websocket: Add support for IPv6."	8 years ago
Richard Mudgett	f997090877	pbx.c: Fix crash from malformed exten pattern. ... Forgetting to indicate an exten is a pattern can cause a crash if the "pattern" has a character set range. e.g., "9999[3-5]" The crash is due to a buffer overwrite because the '-' exten eye-candy wasn't removed as expected and overran the allocated space. The buffer overwrite is fixed two ways in this patch. 1) Fix ext_strncpy() to distinguish between pattern and non-pattern extens. Now '-' characters are removed when they are eye-candy and not when they are part of a pattern character set. Since the function is private to pbx.c, the return value now returns the number of bytes written to the destination buffer instead of the strlen() of the final buffer so the callers that care don't need to add one. 2) Fix callers to ext_strncpy() to supply the correct available buffer size of the destination buffer. ASTERISK-26668 Change-Id: I555d97411140e47e0522684062d174fbe32aa84a	8 years ago
Richard Begg	0dc007e94d	chan_iax2: Reload of iax peer results in loss of host address/port ... When using a non-dynamic peer address, build_peer() invalidates the peer address structure by setting the address family to unspecified. However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup() will not amend the peer address if the cache is still valid, resulting in peer connectivity failures. To fix this, we call ast_dnsmgr_refresh() instead. ASTERISK-26865 Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082	8 years ago
Matt Jordan	59130260e7	configure: Don't use the progress bar with curl when downloading to stdout ... In some scenarios, such as when there may not be a terminal (such as inside a Docker container), curl will apparently direct the progress bar to stdout. This can cause extra data to be appended to a file curl'd down to stdout, resulting in md5 verification failures. This patch removes the progress bar, and tells curl to download the file silently. ASTERISK-26872 #close Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c	8 years ago
zuul	2b611a8d93	Merge "chan_pjsip: Don't assume a session will have a channel."	8 years ago
George Joseph	8470c2bdea	RFC sdp: Initial SDP creation ... * Added additional fields to ast_sdp_options. * Re-organized ast_sdp. * Updated field names to correspond to RFC4566 terminology. * Created allocs/frees for SDP children. * Created getters/setters for SDP children where appropriate. * Added ast_sdp_create_from_state. * Refactored res_sdp_translator_pjmedia for changes. Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48	8 years ago
Joshua Colp	578bc33f6f	Merge "chan_sip: Call not cancelled after receiving a 422 response"	8 years ago
Matt Jordan	05713c36ea	configs/samples/hep.conf.sample: Clarify how the HEP stack works ... This patch updates the documenation in hep.conf.sample to better specify how the various HEP modules interact. ASTERISK-26717 #close Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124	8 years ago
Matt Jordan	0ded269bfa	funcs/func_devstate: Remove new line in Device field of during module load ... During module loading of func_devstate, Asterisk emits the current device state of all Custom device states currently stored in the AstDB. This was erroneously including a new line character ('\n') to the end of the device state, causing two new lines to be emitted in DeviceStateChange AMI events. Note that this only happened for those device state changes that occurred during startup. Regular device state changes for Custom device states are handled elsewhere, and did not have the newline. ASTERISK-26643 #close Reported by: Roman Bedros Tested by: Matt Jordan patches: ami_devstate.diff uploaded by Roman Bedros (License 6842) Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93	8 years ago
Matt Jordan	b03b72717f	main/stasis_cache: Demote the ERROR message when removing a nonexistent item ... This patch demotes the ERROR message that is displayed when a nonexistent item is removed from the Stasis cache. The genesis of this demotion is due to chan_sip's realtime peers and their interaction with Asterisk's core ast_endpoint code, but ostensibly it could happen from other channel drivers as well. Since Mark Michelson already did an excellent job of explaining on this issue, it is quoted here for posterity: "Internally, when a realtime peer is retrieved, Asterisk creates an ast_endpoint structure. When that peer is destroyed, the ast_endpoint is destroyed as well. Part of the destruction of the ast_endpoint involves clearing the Stasis cache of all information about that endpoint. The problem here is that the act of creating the ast_endpoint is not enough to actually put any information in the Stasis cache. Instead, something has to happen, such as a state change, in order for the Stasis cache to have any information about that endpoint. When a device registers, chan_sip creates an ast_endpoint structure, processes the REGISTER, and then destroys the ast_endpoint. When the ast_endpoint is destroyed, there is nothing to destroy in the Stasis cache, so an error message is emitted. When you use rtcachefriends, ast_endpoint structures persist for the lifetime of the module and so you do not see this error message." ASTERISK-25237 #close Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70	8 years ago
Matt Jordan	2d7e68c075	res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue ... Tabs > spaces. Always. Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1	8 years ago
Joshua Colp	12460b05c1	chan_pjsip: Don't assume a session will have a channel. ... When querying for PJSIP specific information using the dialplan function CHANNEL() it is possible that the underlying session will no longer have a channel associated with it. This is most likely to occur when the RTCP HEP module attempts to get the channel name. If this happens then a crash will occur. This change just adds a check that the channel exists on the session before querying it. ASTERISK-26857 Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01	8 years ago
George Joseph	d1ef127084	pjproject_bundled: Reduce the need for rebuilds ... Bundled pjproject should now only rebuild if one of the menuselect "Compiler Flags" options changes. Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43	8 years ago
Joshua Colp	018e01543d	Merge "pjsip/cli_commands: pjsip show channelstats shows wrong codec"	8 years ago
zuul	87aaaef8bb	Merge "res_musiconhold: moh general section is a class and issues warning"	8 years ago
Joshua Colp	9250ddb8f2	Merge "media_cache: Prefer ast_file_is_readable() over access()"	8 years ago
Daniel Journo	36fed72614	pjsip/cli_commands: pjsip show channelstats shows wrong codec ... * cli_commands.c Fixed CLI output ASTERISK-26822 #close Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01	8 years ago
Joshua Colp	6113bb470b	Merge "pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel"	8 years ago
Daniel Journo	b14724adb3	res_musiconhold: moh general section is a class and issues warning ... * res_musiconhold.c: Ensure the general section is not treated as a moh class. ASTERISK-26353 #close Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d	8 years ago
Sean Bright	35cfd2c0cc	media_cache: Prefer ast_file_is_readable() over access() ... Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def	8 years ago
Sean Bright	bc2c66b594	pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel ... Set a variable on the channel that indicates which attempt number we are currently performing to allow for attempt-specific behavior. ASTERISK-26568 #close Reported by: Roman Shubovich Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89	8 years ago
Joshua Colp	4e3b0cedba	res_pjsip_transport_websocket: Add support for IPv6. ... This change adds a PJSIP patch (which has been contributed upstream) to allow the registration of IPv6 transport types. Using this the res_pjsip_transport_websocket module now registers an IPv6 Websocket transport and uses it for the corresponding traffic. ASTERISK-26685 Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647	8 years ago
Daniel Journo	60998371e3	app_voicemail: Cannot set fromstring on a per-mailbox basis ... * apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe	8 years ago
zuul	ff0569740a	Merge "res_http_websocket: Fix faulty read logic."	8 years ago
zuul	9cd1a754a9	Merge "pbx_spool: Gracefully handle long lines in call files"	8 years ago
Mark Michelson	5d0371d743	res_http_websocket: Fix faulty read logic. ... When doing some WebRTC testing, I found that the websocket would disconnect whenever I attempted to place a call into Asterisk. After looking into it, I pinpointed the problem to be due to the iostreams change being merged in. Under certain circumstances, a call to ast_iostream_read() can return a negative value. However, in this circumstance, the websocket code was treating this negative return as if it were a partial read from the websocket. The expected length would get adjusted by this negative value, resulting in the expected length being too large. This patch simply adds an if check to be sure that we are only updating the expected length of a read when the return from a read is positive. ASTERISK-26842 #close Reported by Mark Michelson Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab	8 years ago
Jean Aunis	d51ca4b406	chan_sip: Call not cancelled after receiving a 422 response ... When receiving a 422 response, the invitestate variable must be reset to INV_CALLING. ASTERISK-26841 Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099	8 years ago
Joshua Colp	3ed05badb9	core: Add stream topology changing primitives with tests. ... This change adds a few things to facilitate stream topology changing: 1. Control frame types have been added for use by the channel driver to notify the application that the channel wants to change the stream topology or that a stream topology change has been accepted. They are also used by the indicate interface to the channel that the application uses to indicate it wants to do the same. 2. Legacy behavior has been adopted in ast_read() such that if a channel requests a stream topology change it is denied automatically and the current stream topology is preserved if the application is not capable of handling streams. Tests have also been written which confirm the multistream and non-multistream behavior. ASTERISK-26839 Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9	8 years ago