3701 Commits (b259ac95acefd67a90a451f55acab7fd6ce77d97)

Author	SHA1	Message	Date
Dade Brandon	6dc21bbf00	chan_sip.c: fix websocket_write_timeout default value ... websocket_write_timeout was not being set to its default value during sip config reload, which meant that prior to this commit, 1) the default value of 100 was not used, unless an invalid value (or 1) was specified in sip.conf for websocket_write_timeout, and 2) if the websocket_write_timeout directive was removed from sip.conf without a full restart of asterisk, then the previous value would continue to be used indefinitely. This essentially lead to a 0ms write timeout (the first write attempt in ast_careful_fwrite must have succeeded) in websocket write requests from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf. Changes to websocket_write_timeout still only apply to new websocket sessions, after the sip reload -- timeouts on existing sessions are not adjusted during sip reload. Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953	10 years ago
Joshua Colp	d2c8614122	chan_sip: Enable WebSocket support by default. ... Per the documentation the WebSocket support in chan_sip is supposed to be enabled by default but is not. This change corrects that. Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423	10 years ago
Jonathan Rose	ceebdfce40	chan_sip: Add TCP/TLS keepalive to TCP/TLS server ... Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously this option was only being set on session sockets. http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ According to the link above, the SO_KEEPALIVE option is useful for knowing when a TCP connected endpoint has severed communication without indicating it or has become unreachable for some reason. Without this patch, keep alive is not set on the socket listening for incoming TCP sessions and in Komatsu's report this resulted in the thread listening for TCP becoming stuck in a waiting state. ASTERISK-25364 #close Reported by: Hiroaki Komatsu Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36	10 years ago
Matt Jordan	259fa9c901	Merge "chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c)"	10 years ago
Eugene Voityuk	be693539c3	chan_sip.c: Start ICE negotiation when response is sent or received. ... The current logic for ICE negotiation starts it when receiving an SDP with ICE candidates. This is incorrect as ICE negotiation can only start when each call party have at least one pair of local and remote candidate. Starting ICE negotiation early would result in negotiation failure and ultimately no audio. This change makes it so ICE negotiation is only started when a response with SDP is received or when a response with SDP is sent. ASTERISK-24146 Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca	10 years ago
Filip Jenicek	59a91c350a	chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c) ... Asterisk may crash when calling ast_channel_get_t38_state(c) on a locked channel which is being hung up. ASTERISK-25609 #close Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b	10 years ago
Richard Mudgett	65c8147952	chan_sip: Fix crash involving the bogus peer during sip reload. ... A crash happens sometimes when performing a CLI "sip reload". The bogus peer gets refreshed while it is in use by a new call which can cause the crash. * Protected the global bogus peer object with an ao2 global object container. ASTERISK-25610 #close Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed	10 years ago
Christof Lauber	48c065e46d	chan_sip: Support parsing of Q.850 reason header in SIP BYE and CANCEL requests. ... Current support for reason header did work only in SIP responses. According to RFC3336 the reason header might appear in any SIP request. But it seems to make most sence in BYE and CANCEL so parasing is done there too (if use_q850_reason=yes). Change-Id: Ib6be7b34c23a76d0e98dfd0816c89931000ac790	10 years ago
Richard Mudgett	145d10a5d0	Audit improper usage of scheduler exposed by 5c713fdf18. (v13 additions) ... chan_sip.c: * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to ao2 conversion. * Initialize register scheduler ids earlier because of ASTOBJ to ao2 conversion. chan_skinny.c: * Fix more scheduler usage for the valid 0 id value. ASTERISK-25476 Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95	10 years ago
Richard Mudgett	fa20729032	Audit improper usage of scheduler exposed by 5c713fdf18. ... channels/chan_iax2.c: * Initialize struct chan_iax2_pvt scheduler ids earlier because of iax2_destroy_helper(). channels/chan_sip.c: channels/sip/config_parser.c: * Fix initialization of scheduler id struct members. Some off nominal paths had 0 as a scheduler id to be destroyed when it was never started. chan_skinny.c: * Fix some scheduler id comparisons that excluded the valid 0 id. channel.c: * Fix channel initialization of the video stream scheduler id. pbx_dundi.c: * Fix channel initialization of the packet retransmission scheduler id. ASTERISK-25476 Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8	10 years ago
Steve Davies	d982b99e71	Further fixes to improper usage of scheduler ... When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b	10 years ago
Alexander Traud	cf79b62778	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Corey Farrell	40574a2ea3	chan_sip: Allow websockets to be disabled. ... This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7	10 years ago
Alexander Traud	d343a25173	chan_sip: Do not send all codecs on INVITE. ... Since version 13, Asterisk sent all allowed codecs as callee, even when the caller did not request/support them. In case of dynamic RTP payloads, this led to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the intersection between the requested and the supported codecs is send again. ASTERISK-24543 #close Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287	10 years ago
Alexander Traud	f3b2b3d1b3	chan_sip: Fix autoframing=yes. ... With Asterisk 13, the structures ast_format and ast_codec changed. Because of that, the paketization timing (framing) of the RTP channel moved away from the formats/codecs. In the course of that change, the ptime of the callee was not honored anymore, when the optional autoframing was enabled. ASTERISK-25484 #close Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4	10 years ago
Matt Jordan	13229037d1	channels/chan_sip: Set cause code to 44 on RTP timeout ... To quote Olle: "When issuing a hangup due to RTP timeouts the cause code is not set. I have selected 44 based on Cisco's implementation..." ASTERISK-25135 #close Reported by: Olle Johansson patches: rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267) Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc	10 years ago
Matt Jordan	8cb614fe20	Fix improper usage of scheduler exposed by 5c713fdf18 ... When 5c713fdf18 was merged, it allowed for scheduled items to have an ID of '0' returned. While this was valid per the documentation for the API, it was apparently never returned previously. As a result, several users of the scheduler API viewed the result as being invalid, causing them to reschedule already scheduled items or otherwise fail in interesting ways. This patch corrects the users such that they view '0' as valid, and a returned ID of -1 as being invalid. Note that the failing HEP RTCP tests now pass with this patch. These tests failed due to a duplicate scheduling of the RTCP transmissions. ASTERISK-25449 #close Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39	10 years ago
Walter Doekes	e4df271a3e	chan_sip: Fix From header truncation for extremely long CALLERID(name). ... The CALLERID(num) and CALLERID(name) and other info are placed into the `char from[256]` in initreqprep. If the name was too long, the addr-spec and params wouldn't fit. Code is moved around so the addr-spec with params is placed there first, and then fitting in as much of the display-name as possible. ASTERISK-25396 #close Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260	10 years ago
Matt Jordan	b1f9c998ed	Merge "chan_sip.c: Validation on module reload"	10 years ago
Rodrigo Ramírez Norambuena	34aa96bef4	chan_sip.c: Validation on module reload ... Change validation on reload module because now used the cli function for reload. The sip_reload() function never fail and ever return NULL for this reason on reload() now use the call the sip_reload() and return AST_MODULE_LOAD_SUCCESS. This problem is dectected on reload by PUT method on ARI, getting always 404 http code when the module is reloaded. ASTERISK-25325 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb	10 years ago
Joshua Colp	d03d09aad3	chan_sip: Allow call pickup to set the hangup cause. ... The call pickup implementation in chan_sip currently sets the channel hangup cause to "normal clearing" if call pickup is successfully performed. This action overwrites the "answered elsewhere" hangup cause set by the call pickup code and can result in the SIP device in question showing a missed call when it should not. This change sets the hangup cause to "normal clearing" as a default initially but allows the call pickup to change it as needed. ASTERISK-25346 #close Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff	10 years ago
Richard Mudgett	857923d9c7	chan_sip.c: Set preferred rx payload type mapping on incoming offers. ... ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: I7f04d5c8bee1126fee5fe6afbc39e45104469f4e	10 years ago
Kevin Harwell	43bdddfc26	chan_sip.c: wrong peer searched in sip_report_security_event ... In chan_sip, after handling an incoming invite a security event is raised describing authorization (success, failure, etc...). However, it was doing a lookup of the peer by extension. This is fine for register messages, but in the case of an invite it may search and find the wrong peer, or a non existent one (for instance, in the case of call pickup). Also, if the peers are configured through realtime this may cause an unnecessary database lookup when caching is enabled. This patch makes it so that sip_report_security_event searches by IP address when looking for a peer instead of by extension after an invite is processed. ASTERISK-25320 #close Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4	10 years ago
Alexander Traud	991d4da1eb	chan_sip: Fix negotiation of iLBC 30. ... iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5, only iLBC 30 is negotiated now. ASTERISK-25309 #close Change-Id: I92d724600a183eec3114da0ac607b994b1a793da	10 years ago
Mark Michelson	35a98161df	res_http_websocket: Avoid passing strlen() to ast_websocket_write(). ... We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21 solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67	10 years ago
Richard Mudgett	d122c1e50b	chan_sip.c: Tweak glue->update_peer() parameter nil value. ... Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd	10 years ago
Walter Doekes	34323f9f95	chan_sip: Fix early call pickup channel leak. ... When handle_invite_replaces() was called, and either ast_bridge_impart() failed or there was no bridge (because the channel we're picking up was still ringing), chan_sip would leak a channel. Thanks Matt and Corey for checking the bridge path. ASTERISK-25226 #close Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8	10 years ago
Alexander Traud	1ad9a6b6b6	chan_sip: Reload peer without its old capabilities. ... On reload, previously allowed codecs were not removed. Therefore, it was not possible to remove codecs while Asterisk was running. Furthermore, newly added codecs got appended behind the previous codecs. Therefore, it was not possible to add a codec with a priority of #1. This change removes the old capabilities before the current ones are added. ASTERISK-25182 #close Reported by: Alexander Traud patches: asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520) Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802	10 years ago
Joshua Colp	5caefc98a1	chan_sip: Destroy peers without holding peers container lock. ... Due to the use of stasis_unsubscribe_and_join in the peer destructor it is possible for a deadlock to occur when an event callback is occurring at the same time. This happens because the peer may be destroyed while holding the peers container lock. If this occurs the event callback will never be able to acquire the container lock and the unsubscribe will never complete. This change makes it so the peers that have been removed from the peers container are not destroyed with the container lock held. ASTERISK-25163 #close Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33	10 years ago
Damian Ivereigh	19f60d9412	chan_sip.c: Update dialog fromtag after request with auth ... If a client sends and INVITE which is 401 rejected, then subsequently sends a new INVITE with the auth info and uses a different fromtag from the first INVITE, Asterisk will accept the new INVITE as part of the original dialog - match_req_to_dialog() specifically ignores the fromtag. However it does not update the stored dialog with the new fromtag. This results in Asterisk being unable to match future packets that are part of this dialog (such as the ACK to the OK or the OK to the BYE), and the call is dropped. This problem was originally found when using an NEC-i SV8100-GE (NEC SIP Card). * After a successful match of a packet to the dialog, if the packet is not a SIP_RESPONSE, authentication is present and the fromtags are different, the stored fromtag is updated with the one from the recent INVITE. ASTERISK-25154 #close Reported by: Damian Ivereigh Tested by: Damian Ivereigh Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e	10 years ago
Corey Farrell	80621ce3c5	Fix unsafe uses of ast_context pointers. ... Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa	11 years ago
Corey Farrell	50044fdc15	Stasis: Fix unsafe use of stasis_unsubscribe in modules. ... Many uses of stasis_unsubscribe in modules can be reached through unload. These have been switched to stasis_unsubscribe_and_join. Some subscription callbacks do nothing, for these I've created a noop callback function in stasis.c. This is used by some modules that monitor MWI topics in order to enable cache, since the callback does not become invalid after dlclose it is safe to use stasis_unsubscribe on these, even during module unload. ASTERISK-25121 #close Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c	11 years ago
Rodrigo Ramírez Norambuena	eec010829a	AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro. ... Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723	11 years ago
Joshua Colp	f45833c9ad	Merge "Restrict functionality when ACLs are misconfigured."	11 years ago
Corey Farrell	c3ec5da156	Remove unneeded uses of optional_api providers. ... A few cases exist where headers of optional_api provders are included but not needed. This causes unneeded calls to ast_optional_api_use. * Don't include optional_api.h from sip_api.h. * Move 'struct ast_channel_monitor' to channel.h. * Don't include monitor.h from chan_sip.c, channel.c or features.c. The move of struct ast_channel_monitor is needed since channel.c depends on it. This has no effect on users of monitor.h since channel.h is included from monitor.h. ASTERISK-25051 #close Reported by: Corey Farrell Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478	11 years ago
Mark Michelson	11ffcf662f	Restrict functionality when ACLs are misconfigured. ... This patch has two main purposes: 1) Improve warning messages when ACLs are configured improperly. 2) Prevent misconfigured ACLs from allowing potentially unwanted traffic. To acomplish point (2) in most cases, whatever configuration object that the ACL belonged to was not allowed to load. The one exception is res_pjsip_acl. In that case, ACLs are their own configuration object. Furthermore, the module loading code has no indication that a ACL configuration had a failure. So the tactic taken here is to create an ACL that just blocks everything. ASTERISK-24969 Reported by Corey Farrell Change-Id: I2ebcb6959cefad03cea4d81401be946203fcacae	11 years ago
Corey Farrell	5c1d07baf0	Astobj2: Allow reference debugging to be enabled/disabled by config. ... * The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1	11 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	11 years ago
Kevin Harwell	66f3fd0028	chan_sip: make progressinband default to no ... After the "progressinband" value setting of "never" was updated to never send a 183 this separated its use from the "no" value. Since "never" was the default, but most users probably expect "no" this patch updates the default for the "progressinband" setting to "no." ASTERISK-24835 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4606/ ........ Merged revisions 434654 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	477536ef25	chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled ... When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will attempt to handle both IPv4 and IPv6 addresses, although the information will be stored in a struct with an AF_INET6 address type. However, the current NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly. This patch adds an additional check for the mapped address case, allowing the NAT code to handle clients even when the address is IPv6. Review: https://reviewboard.asterisk.org/r/4563/ ASTERISK-18032 #close Reported by: Christoph Timm patches: nat_with_ipv6.diff submitted by Valentin Vidić (License 6697) ........ Merged revisions 434288 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 434289 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c1cfe3fae2	clang compiler warnings: Fix non-literal-null-conversion warnings ... Clang will flag errors when a char pointer is set to '\0', as opposed to a value that the char pointer points to. This patch fixes this warning in a variety of locations. Review: https://reviewboard.asterisk.org/r/4551 ASTERISK-24917 Reported by: dkdegroot patches: rb4551.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434187 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 434188 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	e48f2e7897	build: Fixes for gcc 5 compilation ... These are fixes for compilation under gcc 5.0... chan_sip.c: In parse_request needed to make 'lim' unsigned. inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99 inline semantics (same as clang). ccss.c: In ast_cc_set_parm, needed to fix weird comparison. dsp.c: Needed to work around a possible compiler bug. It was throwing an array-bounds error but neither sgriepentrog, rmudgett nor I could figure out why. manager.c: In action_atxfer, needed to correct an array allocation. This patch will go to 11, 13, trunk. Review: https://reviewboard.asterisk.org/r/4581/ Reported-by: Jeffrey Ollie Tested-by: George Joseph ASTERISK-24932 #close ........ Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 434114 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ce59fabd5c	clang compiler warnings: Fix invalid enum conversion ... This patch fixes some invalid enum conversion warnings caught by clang. In particular: * chan_sip: Several functions mixed usage of the st_refresher_param enum and st_refresher enum. This patch corrects the functions to use the right enum. * chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state. * strings: Fixed incorrect usage of AO2 flags with strings container. * res_stasis: Change a return enumeration to stasis_app_user_event_res. Review: https://reviewboard.asterisk.org/r/4535 ASTERISK-24917 Reported by: dkdegroot patches: rb4535.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433747 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	d486659502	chan_sip: Simplify dialog/peer references, improve REF_DEBUG output. ... * Replace functions for ref/undef of dialogs and peers with macro's to call ao2_t_bump/ao2_t_cleanup. * Enable passthough of REF_DEBUG caller information to sip_alloc and find_call. ASTERISK-24882 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4189/ ........ Merged revisions 433115 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	2c83ac4364	chan_sip: Fix dialog reference leaked to scheduler for reinvite_timeout. ... Release the scheduler reference to the dialog for reinvite timeout during dialog_unlink_all. ASTERISK-24876 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4491/ ........ Merged revisions 433112 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433113 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	c08fd275bf	Logger: Convert 'struct ast_callid' to unsigned int. ... Switch logger callid's from AO2 objects to simple integers. This helps in two ways. Copying integers is faster than referencing AO2 objects, so this will result in a small reduction in logger overhead. This also erases the possibility of an infinate loop caused by an invalid callid in threadstorage. ASTERISK-24833 #comment Committed callid conversion to trunk. Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f1ab2c5e8b	chan_sip: Fix realtime locking inversion when poking a just built peer. ... When a realtime peer is built it can cause a locking inversion when the just built peer is poked. If the CLI command "sip show channels" is periodically executed then a deadlock can happen because of the locking inversion. * Push the peer poke off onto the scheduler thread to avoid the locking inversion of the just built realtime peer. AST-1540 ASTERISK-24838 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4454/ ........ Merged revisions 432526 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432528 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3725173b9e	channels/chan_sip: Don't send a BYE after final response when PBX thread fails ... When Asterisk fails to start a PBX thread for a new channel - for example, when the maxcalls setting in asterisk.conf is exceeded - we currently send a final response, and then attempt to send a BYE request to the UA. Since that's all sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt such that we don't get stuck sending BYE requests to something that does not want it. Note that this patch is a slight modification of the one on ASTERISK-15434. For clarity, it explicitly calls sipalreadygone with the calls to transmit a final response. ASTERISK-21845 ASTERISK-15434 #close Reported by: Makoto Dei Tested by: Matt Jordan patches: sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027) ........ Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432321 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4b63da7f7d	channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario ... When we receive an SDP as part of an offer/answer for a peer/friend has been configured to require encryption, and that SDP offer/answer failed to provide acceptable crypto attributes, we currently issue a WARNING that uses the phrase "we" and "requested". In this case, both of those terms are ambiguous - the user will probably think "we" is Asterisk (it most likely isn't) and it may not be a "request", so much as an SDP that was received in some fashion. This patch makes the WARNING messages slightly less bad and a bit more accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton ........ Merged revisions 432277 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432278 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8574c4d197	channels/chan_sip: Fix crash when transmitting packet after thread shutdown ... When the monitor thread is stopped, its pthread ID is set to a specific value (AST_PTHREADT_STOP) so that later portions of the code can determine whether or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit failed to check for that value, checking instead only for AST_PTHREAD_STOP. Passing the invalid yet very specific value to pthread_kill causes a crash. This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that it doesn't attempt to poke the thread if the thread has already been stopped. ASTERISK-24800 #close Reported by: JoshE ........ Merged revisions 432198 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432199 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago