8245 Commits (a6b16d7029a7a44c6a43b3b0f1cbc74cecd11454)

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	10 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	10 years ago
Joshua Colp	1e41d14822	Merge "chan_sip: Make autocreated peers send PeerStatus events"	10 years ago
Joshua Colp	c480159045	chan_sip: Give more time for TCP/TLS threads to stop. ... The unload process currently tells each TCP/TLS to terminate but does not wait for them to do so. This introduces a race condition where the container holding the threads may be destroyed before the threads are able to remove themselves from it. When they finally do the container is invalid and can't be used causing a crash. A previous change existed which waited a bit to wait for any stranglers to finish. This change extends this and waits longer. ASTERISK-25961 #close Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6	10 years ago
kkm	41ecf22587	chan_sip: Make autocreated peers send PeerStatus events ... Since Stasis has been introduced, an attempt to send AMI messages by an autocreated peer caused a crash, and all events from autocreated peers were semi-inadvertently disabled altogether in 0b83761. This change restores the disabled functionality. ASTERISK-25950 Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974	10 years ago
Jaco Kroon	f06ce7f90a	chan_sip: Don't verify table if rtupdate=no ... If rtupdate=no do not verify sipregs/peers table has updatable fields. ASTERISK-25934 #close Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d	10 years ago
Richard Mudgett	6138a75e8e	pbx.h: Make ast_state_cb_type take more const. ... This eliminates some casts that I made a note saying v10 and above would no longer need them. Better late than never :) Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572	10 years ago
zuul	a04a62586f	Merge "chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers."	10 years ago
George Joseph	c4064727d2	chan_pjsip: Add 'pjsip show channelstats' ... Added the ability to show channel statistics to chan_pjsip (cli_functions.c) Moved the existing 'pjsip show channel(s)' functionality from pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's private header so it made sense to move the existing channel commands as well. Now using stasis_cache_dump to get the channel snapshots rather than retrieving all endpoints, then getting each one's channel snapshots. Much more efficient. Change-Id: I03b114522126d27434030b285bf6d531ddd79869	10 years ago
Alexander Traud	3f720155b7	chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. ... Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those codecs, which the caller did not request/support. That fix was not complete because on the second Session Timer all codecs were sent again. Some VoIP/SIP clients interpreted that complete codec-list as a change in the SIP session. Because of that, Asterisk did not send the RTP audio via NAT anymore which created a non-audio scenario after the second Session Timer fired. ASTERISK-24543 #close Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66	10 years ago
Francesco Castellano	ac66999971	chan_sip.c: Space after port causes unnecessary resolution attempt ... check_via() already skips leading blanks where the sent-by address (with the optional port) should be placed. Since RFC 3261 allows for blanks between the port ant the Via parameters: > https://tools.ietf.org/html/rfc3261#section-20.42 (actually it allows a lot of blanks more ;-)). I just switched from ast_skip_blanks() to ast_strip() on the local copy of the string. ASTERISK-21301 #close Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06	10 years ago
Kevin Harwell	a3c9a74a02	chan_pjsip: ref leak when checking direct_media_glare ... Fix the reference leak introduced in the following commit: c534bd5807 ASTERISK-25849 Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85	10 years ago
zuul	4aaf8f2ace	Merge "chan_pjsip: transfers with direct media reinvite has wrong address/port"	10 years ago
Kevin Harwell	c534bd5807	chan_pjsip: transfers with direct media reinvite has wrong address/port ... During a transfer involving direct media a race occurs between when the transferer channel is swapped out, initiating rtp changes/updates, and the subsequent reinvites. When Alice, after speaking with Charlie (Bob is on hold), connects Bob and Charlie invites are sent to each in order to establish the call between them. Bob is taken off hold and Charlie is told to have his media flow through Asterisk. However, if before those invites go out the bridge updates Bob's and/or Charlie's rtp information with direct media data (i.e. address, port) then the invite(s) will contain the remote data in the SDP instead of the Asterisk data. The race occurs in the native bridge glue code when updating the peer. The direct_media_address can get set twice before sending out the first invite during call connection. This can happen because the checking/setting of the direct_media_address happened in one thread while the sending of the invite(s) happened in another thread. This fix removes the race condition by moving the checking/setting of the direct_media_address to be in the same thread as the sending of the invites(s). This serializes the checking/setting and sending so they can no longer happen out of order. ASTERISK-25849 #close Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873	10 years ago
Joshua Colp	7d63b4499e	Merge "chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full()."	10 years ago
Richard Mudgett	810f92c9dc	chan_sip.c: Fix mwi resub deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 #close Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6	10 years ago
Richard Mudgett	72c444ba37	chan_sip.c: Fix registration timeout and expire deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508	10 years ago
Richard Mudgett	7ea1e181dc	chan_sip.c: Fix waitid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Made always run check_pendings() under the scheduler thread so scheduler ids can be checked safely. ASTERISK-25023 Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52	10 years ago
Richard Mudgett	fbf8e04aed	chan_sip.c: Fix t38id deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f	10 years ago
Richard Mudgett	02458cc6fd	chan_sip.c: Fix session timers deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900	10 years ago
Richard Mudgett	c7fdff2e37	chan_sip.c: Fix reinviteid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1	10 years ago
Richard Mudgett	69810b306d	chan_sip.c: Fix autokillid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix clearing autokillid in __sip_autodestruct() even though we could reschedule. ASTERISK-25023 Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f	10 years ago
Richard Mudgett	f484ddbdfe	chan_sip.c: Fix packet retransid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix retrans_pkt() to call check_pendings() with both the owner channel and the private objects locked as required. * Refactor dialog retransmission packet list to safely remove packet nodes. The list nodes are now ao2 objects. The list has a ref and the scheduled entry has a ref. ASTERISK-25023 Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641	10 years ago
Richard Mudgett	67c79c326d	chan_sip.c: Fix provisional_keepalive_sched_id deadlock. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48	10 years ago
Richard Mudgett	76be7093cd	chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events. ... This patch is part of a series to resolve deadlocks in chan_sip.c. * Make dialog_unlink_all() unschedule all items at once in the sched thread. ASTERISK-25023 Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4	10 years ago
Richard Mudgett	52f0932e4c	chan_sip.c: Clear scheduled immediate events on unload. ... This patch is part of a series to resolve deadlocks in chan_sip.c. The reordering of chan_sip's shutdown is to handle any immediate events that get put onto the scheduler so resources aren't leaked. The typical immediate events at this time are going to be concerned with stopping other scheduled events. ASTERISK-25023 Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20	10 years ago
Richard Mudgett	0987a11cce	sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Delaying destruction of the chan_sip sip_pvt structures caused the /channels/chan_sip/test_sip_rtpqos unit test to crash. That test registers a special test ast_rtp_engine with the rtp engine module. When the unit test completes it cleans up by unregistering the test ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt happens after the unit test returns, the destructor tries to call the rtp engine destroy callback of the test ast_rtp_engine auto variable which no longer exists on the stack. * Change the test ast_rtp_engine auto variable to a static variable. Now the variable can still exist after the unit test exits so the delayed sip_pvt destruction can complete successfully. ASTERISK-25023 Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13	10 years ago
Richard Mudgett	cb97198ca6	chan_sip.c: Simplify sip_pvt destructor call levels. ... Remove destructor calling destroy_it calling really_destroy_it for no benefit. Just make the destructor the really_destroy_it function. Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a	10 years ago
Richard Mudgett	8be01398d9	chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full(). ... Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12	10 years ago
George Joseph	d2eb65f71e	res_pjsip: Strip spaces from items parsed from comma-separated lists ... Configurations like "aors = a, b, c" were either ignoring everything after "a" or trying to look up " b". Same for mailboxes, ciphers, contacts and a few others. To fix, all the strsep(&copy, ",") calls have been wrapped in ast_strip. To facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were updated to handle null pointers. In some cases, an ast_strlen_zero() test was added to skip consecutive commas. There was also an attempt to ast_free an ast_strdupa'd string in ast_sip_for_each_aor which was causing a SEGV. I removed it. Although this issue was reported for realtime, the issue was in the res_pjsip modules so all config mechanisms were affected. ASTERISK-25829 #close Reported-by: Mateusz Kowalski Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2	10 years ago
Richard Mudgett	25de01f301	SIP diversion: Fix REDIRECTING(reason) value inconsistencies. ... Previous chan_sip behavior: Before this patch chan_sip would always strip any quotes from an incoming reason and pass that value up as the REDIRECTING(reason). For an outgoing reason value, chan_sip would check the value against known values and quote any it didn't recognize. Incoming 480 response message reason text was just assigned to the REDIRECTING(reason). Previous chan_pjsip behavior: Before this patch chan_pjsip would always pass the incoming reason value up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip would send the reason value as passed down. With this patch: Both channel drivers match incoming reason values with values documented by REDIRECTING(reason) and values documented by RFC5806 regardless of whether they are quoted or not. RFC5806 values are mapped to the equivalent REDIRECTING(reason) documented value and is set in REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a quoted string version ('"unconditional"') is converted to REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal with 'cfu' instead of any of the aliases. The incoming 480 response reason text supported by chan_sip checks for known reason values and if not matched then puts quotes around the reason string and assigns that to REDIRECTING(reason). Both channel drivers send outgoing known REDIRECTING(reason) values as the unquoted RFC5806 equivalent. User custom values are either sent as is or with added quotes if SIP doesn't allow a character within the value as part of a RFC3261 Section 25.1 token. Note that there are still limitations on what characters can be put in a custom user value. e.g., embedding quotes in the middle of the reason string is silly and just going to cause you grief. * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the 'cfu' value. * Added missing malloc() NULL return check in res_pjsip_diversion.c set_redirecting_reason(). * Fixed potential read from a stale pointer in res_pjsip_diversion.c add_diversion_header(). The reason string needed to be copied into the tdata memory pool to ensure that the string would always be available. Otherwise, if the reason string returned by reason_code_to_str() was a user's reason string then the string could be freed later by another thread. Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87	10 years ago
Richard Mudgett	2dae4a1ccf	chan_sip.c: Fix T.38 issues caused by leaving a bridge. ... chan_sip could not handle AST_T38_TERMINATED frames being sent to it when the channel left the bridge. The action resulted in overlapping outgoing reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not happy. * Force T.38 to be remembered as locally bridged. Now when the channel leaves the native RTP bridge after T.38, the channel remembers that it has already reINVITEed the media back to Asterisk. It just needs to terminate T.38 when the AST_T38_TERMINATED arrives. * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if they happen before the T.38 state changes to disabled. Now the T.38 state is set to disabled before the reINVITE is sent. ASTERISK-25582 #close Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce	10 years ago
zuul	170941990b	Merge "chan_sip: Optionally supply fromuser/fromdomain in SIP dial string."	10 years ago
Richard Mudgett	886ee09471	chan_sip.c: Suppress T.38 SDP c= line if addr is the same. ... Use the correct comparison function since we only care if the address without the port is the same. Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0	10 years ago
Walter Doekes	c00082329e	chan_sip: Optionally supply fromuser/fromdomain in SIP dial string. ... Previously you could add [!dnid] to the SIP dial string to alter the To: header. This change allows you to alter the From header as well. SIP dial string extra options now look like this: [![touser[@todomain]][![fromuser][@fromdomain]]] INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To: header, that is no longer possible. ASTERISK-25803 #close Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7	10 years ago
George Joseph	b69729dde5	chan_misdn: Fix a few issues causing compile errors ... Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98	10 years ago
Richard Mudgett	a877e0d94b	AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow. ... Setting the sip.conf timert1 value to a value higher than 1245 can cause an integer overflow and result in large retransmit timeout times. These large timeout times hold system file descriptors hostage and can cause the system to run out of file descriptors. NOTE: The default sip.conf timert1 value is 500 which does not expose the vulnerability. * The overflow is now detected and the previous timeout time is calculated. ASTERISK-25397 #close Reported by: Alexander Traud Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290	10 years ago
Mark Michelson	d89f0b09de	Merge "chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip."	10 years ago
George Joseph	40da6434c1	build_system: Fix some warnings highlighted by clang ... Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd	10 years ago
StefanEng86	55a7367ad4	chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip. ... When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a) AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect asterisk to include the same value for its own ip in both cases a) and b), but it seems a) produces a contact header like Contact: <sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like <sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf My guess is that manager_sipnotify should call ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does, because after applying this patch, both cases a) and b) produce the contact header that I expect: <sip:asterisk@192.168.1.227:8060> Reported by: Stefan Engström Tested by: Stefan Engström Change-Id: I86af5e209db64aab82c25417de6c768fb645f476	10 years ago
Corey Farrell	830f8933c2	chan_sip: Fix buffer overrun in sip_sipredirect. ... sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer of 256 characters. This patch reduces the copy to 255 characters to leave room for the string null terminator. ASTERISK-25722 #close Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab	10 years ago
Mark Michelson	53570e2c6f	Merge "chan_sip: option 'notifyringing' change and doc fix"	10 years ago
Ward van Wanrooij	d4b10cfb3e	chan_sip: option 'notifyringing' change and doc fix ... In the sample sip.conf this is written with regard to notifyringing: ;notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) However, this setting changes whether or not any RINGING indications are sent to subscriptions. There is no separate configurable setting that allows to control whether INUSE subscriptions also get sent RINGING. This is however a useful option, to see (using BLF) if somebody else is able to handle an incoming call or if everybody is busy. This patch corrects the documentation for notifyringing (so the documentation matches the functionality) and make notifyringing a tri-state option, by adding the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing = notinuse, only subscriptions that are not INUSE are sent the RINGING signal. The default setting for notifyringing remains set to yes, so the default behaviour is not affected. ASTERISK-25558 Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa	10 years ago
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