3764 Commits (79532bca7526e6652608d5a8f11aadf98003e946)

Author	SHA1	Message	Date
zuul	3f62485ba7	Merge "chan_sip: Resolve externhost not to IPv6; instead go for IPv4."	9 years ago
Alexander Traud	5dd99465d3	chan_sip: Resolve externhost not to IPv6; instead go for IPv4. ... For the channel driver chan_sip, you specify externhost=example.com in sip.conf when your Asterisk is behind a NAT and your IP address is assigned dynamically. Or stated differently: You do not have a static IP address to use "externaddr" directly. This NAT support is quite handy but just about IPv4. Previously, Asterisk resolved "externhost" to any IP version. When the first DNS answer resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and connection (c=). This happened in outgoing SIP-REGISTER and while answering SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". ASTERISK-18232 #close Reported by: Jacek Kowalski Tested by: Alexander Traud patches: changes.patch submitted by Alessandro Crespi Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac	9 years ago
George Joseph	d425971009	chan_sip: Address runaway when realtime peers subscribe to mailboxes ... Users upgrading from asterisk 13.5 to a later version and who use realtime with peers that have mailboxes were experiencing runaway situations that manifested as a continuous stream of taskprocessor congestion errors, memory leaks and an unresponsive chan_sip. A related issue was that setting rtcachefriends=no NEVER worked in asterisk 13 (since the move to stasis). In 13.5 and earlier, when a peer tried to register, all of the stasis threads would block and chan_sip would again become unresponsive. After 13.5, the runaway would happen. There were a number of causes... * mwi_event_cb was (indirectly) calling build_peer even though calls to mwi_event_cb are often caused by build_peer. * In an effort to prevent chan_sip from being unloaded while messages were still in flight, destroy_mailboxes was calling stasis_unsubscribe_and_join but in some cases waited forever for the final message. * add_peer_mailboxes wasn't properly marking the existing mailboxes on a peer as "keep" so build_peer would always delete them all. * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions then just creating them again. All of this was causing a flood of subscribes and unsubscribes on multiple threads all for the same peer and mailbox. Fixes... * add_peer_mailboxes now marks mailboxes correctly and build_peer only deletes the ones that really are no longer needed by the peer. * add_peer_mwi_subs now only adds subscriptions marked as "new" instead of unsubscribing and resubscribing everything. It also adds the peer object's address to the mailbox instead of its name to the subscription userdata so mwi_event_cb doesn't have to call build_peer. With these changes, with rtcachefriends=yes (the most common setting), there are no leaks, locks, loops or crashes at shutdown. rtcachefriends=no still causes leaks but at least it doesn't lock, loop or crash. Since making rtcachefriends=no work wasnt in scope for this issue, further work will have to be deferred to a separate patch. Side fixes... * The ast_lock_track structure had a member named "thread" which gdb doesn't like since it conflicts with it's "thread" command. That member was renamed to "thread_id". ASTERISK-25468 #close Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0	9 years ago
zuul	544fe73811	Merge "chan_sip: Fix session timeout on retransmit of non-UDP packets"	9 years ago
Steve Davies	6ba68b486e	chan_sip: Fix session timeout on retransmit of non-UDP packets ... Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP connections, allowing the TCP layer to handle the retransmits. Unfortunately, this caused sessions to be terminated with a retransmit timeout becasue it stopped at the point of the first retrans call. This patch waits for the 64*T1 timer to expire instead. ASTERISK-19968 Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204	9 years ago
Walter Doekes	740292e6ae	chan_sip: Allow target refresh (Contact update) on re-INVITE. ... Previously, the Contact was stored only on initial INVITE and on any 18X and 200. That meant that after re-INVITEs from *us* the Contact could get updated, but after re-INVITEs from the *peer*, it did not. This changeset fixes this inconsistency, properly allowing target refreshes through re-INVITES (RFC3261, 12.2). If your strictrtp setting allows it, this change allows you to switch the source IP of a connected/calling device mid-call with a simple re-INVITE from the new IP. ASTERISK-26358 #close Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435	9 years ago
Joshua Colp	82a3d659dc	chan_sip: Don't allocate new RTP instances on top of old ones. ... In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73	9 years ago
Alexander Traud	7a12355dbd	chan_sip: Allow Preferred sRTP. ... Following the Encrypt-all-the-things paradigm: The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile (sRTP is preferred aka optional; not mandatory). If the VoIP server does not support sRTP and TLS, the phone shows an open padlock icon. This paradigm is supported by several VoIP/SIP clients on default. Some implementations even cannot be changed to RTP/sAVP. Therefore here, this change allows Preferred sRTP for ingress. For egress, please, create a dial plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP. ASTERISK-20234 #close Reported by: tootai Tested by: tootai, Alexander Traud patches: srtp_patches.diff submitted by Matt Jordan Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd	9 years ago
Walter Doekes	d80b28560c	chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. ... Certain SNOM phones send so-called "optional crypto" in their SDP body. Regular SRTP setup looks like this: m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... SNOM-style "optional crypto" looks like this: m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... A crypto line is supplied, but the m-line does not have SAVP. When res_srtp.so is *not* loaded, then chan_sip.so treats the optional crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the incoming call with the following message: WARNING: process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio For platforms that want to start providing SRTP this presents a compatibility problem. This changeset lets chan_sip handle the SDP as if no crypto-line was supplied: i.e. accept the call as regular RTP, just like it did before res_srtp was loaded. Now you'll get this informative warning instead: WARNING: Ignoring crypto attribute in SDP because RTP transport is insecure ASTERISK-23989 #close Reported by: Olle Johansson Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2	9 years ago
Kevin Harwell	53a2f7dc88	res_format_attr_g729: Add annexb=no format parameter to SDPs ... Historically, Asterisk has always specified annexb=no for the g729 format. However, when using res_pjsip no format attribute was specified. This patch makes it so the SDP now contains a format attribute line with annexb=no. Note, that this means only g729a is negotiated. Even for pass through support. According to rfc7261 the type of annex used (a or b) is dependent upon the answerer. However, Asterisk being a back to back user agent makes this tricky to support at this time, thus we only allow annex 'a' for now. ASTERISK-26228 #close patches: res_format_attr_g729.c submitted by Jason Parker (license 4993) Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0	9 years ago
Corey Farrell	824a4e84d1	Refactor usage pattern of xmldoc info tag. ... This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea	9 years ago
cjack	957df73301	chan_sip: Fix lastrtprx always updated ... Packets are read regulary, when there is no data in buffer fr->frametype is AST_FRAME_NULL. There was no check of frametype and lastrtprx always updated and, therefore, rtptimeout did not work at all. ASTERISK-25270 #close Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d	9 years ago
zuul	7cfd9bf104	Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."	9 years ago
Alexander Traud	9be69c1636	chan_sip: Enable Session-Timers for SIP over TCP (and TLS). ... Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957	9 years ago
zuul	075f7c4aea	Merge "chan_sip: Prevent deadlock when issuing "sip show channels""	9 years ago
zuul	9473818659	Merge "res_srtp: Enable AES-256 and AES-GCM."	9 years ago
George Joseph	1b4922466b	chan_sip: Prevent deadlock when issuing "sip show channels" ... sip_show_channels locks the dialogs container first then locks each sip_pvt so it can spit out the details. The rest of sip dialog processing locks the sip_pvt first then locks the dialogs container if it needs to. Both lock in the order they need but deadlocks can result. To fix, sip_show_channels and sip_show_channelstats have been converted to use an iterator rather than ao2_callback. This way the container is locked only while getting the next entry and is unlocked when the callback is called. ASTERISK-23013 #close Change-Id: Id9980419909e811f89484950ed46ef117b9eb990	9 years ago
Alexander Traud	1d2173c7ae	res_srtp: Enable AES-256 and AES-GCM. ... ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b	9 years ago
Richard Mudgett	db4979fa79	chan_sip.c: Fix deadlock potential in fax redirection. ... The sip_read() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e	9 years ago
Corey Farrell	f73ddde7d4	chan_sip: Fix reference leak in mwi_event_cb ... Cleanup the peer reference when stasis_subscription_final_message is true. Also free peer_name even if peer exists, after reload a new peer_name will be allocated. ASTERISK-26193 #close Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69	9 years ago
Corey Farrell	ad30d60c69	chan_sip: Fix reference leaks in error paths. ... * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error. * build_peer leaks peer on failure to allocate the endpoint. This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed with an unref in the appropriate place. ASTERISK-26184 #close Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12	9 years ago
Joshua Colp	302be4809a	chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. ... Some T.38 implementations may send another re-invite after the initial one which adds additional negotiation details (such as the max bitrate). Currently this will fail when passthrough is being done in chan_sip as we do nothing if T.38 is already active. Other handlers of T.38 inside of Asterisk (such as res_fax) handle this scenario so this change adds support for it to chan_sip and res_pjsip_t38. If a request to negotiate is received while T.38 is already enabled a new re-INVITE is sent and negotiation is done again. ASTERISK-26179 #close Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c	9 years ago
Joshua Colp	75818b4084	siren: Add format attribute modules for Siren7 and Siren14. ... This change removes hardcoded SDP parsing and generation for Siren7 and Siren14 from chan_sip and moves it to format attribute modules so it can also be used by chan_pjsip. With this the fmtp lines for both are added with the bitrate information. ASTERISK-26021 Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037	9 years ago
zuul	947f76a971	Merge "chan_sip: bigger buffers for headers, better failure mode"	9 years ago
zuul	88dfcd21b2	Merge "chan_sip: Support auth username for callbackextension feature"	9 years ago
George Joseph	d21a77b325	build: Fix ast_sockaddr initialization to be more portable ... A change to glibc 2.22 changed the order of the sockadddr_storage members which caused the places where we do an initialization of ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those initializers (which we shouldn't have been using anyway) have been replaced with memsets. Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4	9 years ago
Joshua Colp	2525563438	Merge "chan_sip: No rtpmap for static RTP payload IDs in SDP."	9 years ago
Timo Teräs	39b69ab537	Fixes to include signal.h ... POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc	9 years ago
Alexander Traud	784c18128b	chan_sip: No rtpmap for static RTP payload IDs in SDP. ... This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compactheaders=yes via the file sip.conf. ASTERISK-25578 #close Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044	9 years ago
Vasil Kolev	80ff7912a1	chan_sip: bigger buffers for headers, better failure mode ... Currently chan_sip can give weird messages if the contacts don't fit in the From: or To: headers. This fix changes the from,to and invite variables to use ast_str, allocates and deallocates them and resizes them if needed. ASTERISK-26069 #close Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3	9 years ago
Timo Teräs	538c6415c6	chan_sip: Support auth username for callbackextension feature ... ASTERISK-20527 #close Change-Id: I659cf7f00836a09d09d146ad226a40477d731239	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
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
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	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	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	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	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	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	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	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