asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	91513c5e8d	Merge "channels/chan_pjsip: fix HANGUPCAUSE function bug."	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
Joshua Colp	8d5219e6b8	Merge "core: Ensure presencestate subtype and message are NULL."	9 years ago
Joshua Colp	cdf2522160	Merge "res_odbc: Make pooling option deprecation notice more useful."	9 years ago
Joshua Colp	8966c8ec5a	Merge "cdr_mysql: fix UTC support"	9 years ago
Aaron An	18a8ca06eb	channels/chan_pjsip: fix HANGUPCAUSE function bug. HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. This patch change the call order of ast_queue_control_data and ast_queue_control in chan_pjsip_incoming_response. ASTERISK-26396 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Ide2d31723d8d425961e985de7de625694580be61	9 years ago
zuul	9ef0eb6487	Merge "logger: Simplify ast_callid handling code."	9 years ago
Joshua Colp	57b29f3b69	Merge "logger: Always enable verbose for console channel."	9 years ago
Joshua Colp	a805d779e8	core: Ensure presencestate subtype and message are NULL. When retrieving presence state information there is no guarantee that the subtype and message passed in are set to NULL. This change ensures they are. ASTERISK-26397 #close Change-Id: If38cd730e409e9a9b6eb9adef6591d15a9e61f86	9 years ago
zuul	ccc0bfa69c	Merge "logger: Fix default console settings."	9 years ago
zuul	4caee4a11b	Merge "core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get."	9 years ago
Joshua Colp	077caf566e	res_odbc: Make pooling option deprecation notice more useful. This changes the notice for the deprecation of the old pooling options to point to the new option for doing pooling. This gives a clearer direction as to what to look into. ASTERISK-26389 #close Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10	9 years ago
Joshua Colp	78b6190a11	odbc: Remove options that are no longer applicable. The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6	9 years ago
zuul	f84652bd81	Merge "asterisk.c: Non-root users also get the astcanary after core restart."	9 years ago
Corey Farrell	923edf2596	logger: Simplify ast_callid handling code. Routines responsible for managing ast_callid's are overly complicated. This is left-over code from when ast_callid was an AO2 object. Now that it is an integer the code can be reduced. ast_callid handler code no longer prints it's own error message upon failure to allocate threadstorage as ast_calloc would have already printed a message. Debug messages that were printed when TEST_FRAMEWORK was enabled have been also been removed. Change-Id: I65a768a78dc6cf3cfa071e97f33ce3dce280258e	9 years ago
Corey Farrell	5cb905a227	core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get. Move the function outside the conditional block that excludes LOW_MEMORY. ASTERISK-26273 #close Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4	9 years ago
zuul	0df2373434	Merge "res_pjsip_multihomed: Change Contact port to listening port."	9 years ago
Corey Farrell	00f1d05d34	logger: Always enable verbose for console channel. Previous versions of Asterisk did not require verbose to be specified in logger.conf for the console channel, if it was requested by command line or asterisk.conf it just worked. This change causes Asterisk to always enable verbose in the console channel level mask. Verbose is displayed on consoles if requested by command line, option_verbose or 'core set verbose'. This also delays initialization of the logger until after threadstorage is initialized. Initializing too early can cause messages to be printed multiple times to the console (stdout). ASTERISK-26391 #close Change-Id: I52187d67c2fcb3efd5561bf04b3e5e23e5ee8a04	9 years ago
Corey Farrell	74f562a8e2	logger: Fix default console settings. When logger.conf is missing or invalid we should be printing notices, warnings and errors to the console. The logmask was incorrectly calculated. Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3	9 years ago
zuul	ea8105cf5e	Merge "sd_notify (systemd status notifications) support"	9 years ago
zuul	0caf846aff	Merge "rtp: Only accept the first payload for a format in SDP."	9 years ago
zuul	d36da3a26b	Merge "Fix showing of swap details when sysinfo() is available"	9 years ago
Walter Doekes	0bc9912739	asterisk.c: Non-root users also get the astcanary after core restart. Without this change, a 'core restart' would kill the astcanary forever if you're not running as root. Both with and without this patch, the scheduling priority was still SCHED_RR after restart. Additionally, the astcanary is now spawned if you start with high priority and Asterisk doesn't get a chance to lower it. For example through: `chrt -r 10 sudo -u asterisk asterisk -c` Also reap killed astcanary processes on core restart. ASTERISK-26352 #close Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55	9 years ago
zuul	5f9ad3e57e	Merge "res_config_odbc.c: Fix buffer size limitation creating invalid SQL."	9 years ago
zuul	2360cd3ed2	Merge "asterisk.c: When astcanary dies on linux, reset priority on all threads."	9 years ago
Walter Doekes	bffaf46690	asterisk.c: When astcanary dies on linux, reset priority on all threads. Previously only the canary checking thread itself had its priority set to SCHED_OTHER. Now all threads are traversed and adjusted. ASTERISK-19867 #close Reported by: Xavier Hienne Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39	9 years ago
Richard Mudgett	2820b13393	res_config_odbc.c: Fix buffer size limitation creating invalid SQL. Creating ODBC SQL queries resulted in queries too large to fit into the supplied buffer. The resulting truncated buffer contained an invalid SQL query. * Made SQL query generation code use a thread storage buffer that can increase in size as needed. * Fixed bad multi-line warning messages. ASTERISK-26263 #close Reported by: Jeppe Ryskov Larsen Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae	9 years ago
Joshua Colp	0376af9519	rtp: Only accept the first payload for a format in SDP. When receiving an SDP offer with multiple payloads for the same format we would generate an answer with the first payload, but during the payload crossover operation (to set the payloads for receiving) we would remove all payloads but the last. This would result in incoming traffic being matched against the wrong format and outgoing traffic being sent using the wrong payload. This change makes it so that once a format has a payload number put into the mapping all subsequent ones are ignored. This ensures there is only ever one payload in the mapping and that it is the payload placed into the answer SDP. ASTERISK-26365 #close Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60	9 years ago
Joshua Colp	9d894ee0a1	res_pjsip_multihomed: Change Contact port to listening port. The res_pjsip_multihomed module determines what interface and transport a request is going out on and updates the SIP message accordingly with the address information. This currently incorrectly updates the Contact header for connectionful protocols to the ephemeral connection port, instead of the bound address for the listening socket which can actually accept the connection back. If the remote side attempts to connect back on the epehemeral port it will fail. This change makes it so the port is updated to the bound port on connectionful protocols and is maintained on UDP (as there can be multiple of those). ASTERISK-26374 #close Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab	9 years ago
George Joseph	47c527df0a	pjproject_bundled: Prevent SERVFAIL from marking name server bad A name server that returns "Server Failure" is indicating only that the server couldn't process that particular request. We should NOT assume that the name server is incapable of serving other requests. Here's the scenario we've been encountering... * 2 local name servers configured in resolv.conf. * An OPTIONS request causes a request for A and AAAA records to go out to both nameservers. * The A responses both come back successfully resolved. * Because of an issue at some upstream nameserver, the AAAA responses for that particular query come back as "SERVFAIL" from both local name servers. * Both local servers are marked as bad and no further queries can be sent until the 60 second ttl expires. Only previously cached results can be used. * In this case, 60 seconds is just enough time for another OPTIONS request to go out to the same host so the cycle repeats. We could set the bad ttl really low but that also affects REFUSED and NOTAUTH which probably DO signal a real server issue. Besides, even a really low bad ttl would be an issue on a pbx. Although we use our own resolver in 14 and master and don't have this issue there, Teluu has merged this patch upstream so it's appropriate to cherry-pick to 14 and master to keep pjproject consistent. Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0	9 years ago
Tzafrir Cohen	d3ddf4b0fd	cdr_mysql: fix UTC support * Make 'cdrzone=UTC' work properly. * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone ASTERISK-26359 #close Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778	9 years ago
Tzafrir Cohen	07b95f7c65	sd_notify (systemd status notifications) support sd_notify() is used to notify systemd of changes to the status of the process. This allows the systemd daemon to know when the process finished loading (and thus only start another program after Asterisk has finished loading). To use this, use a systemd unit with 'Type=notify' for Asterisk. This commit also adds the function ast_sd_notify(), a wrapper around sd_notify that does nothing if not built with systemd support. Also adds support for libsystemd detection in the configure script. Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811	9 years ago
Timo Teräs	bc81765bb4	Fix showing of swap details when sysinfo() is available If sysinfo() is available, but not sysctl() or swapctl() the printing code for swap buffer sizes is incorrectly omitted. The above condition happens with musl c-library. Fix #if rule to consider defined(HAVE_SYSINFO). And also remove the redundant \|\| defined(HAVE_SYSCTL) which was incorrectly there to start with. Now swap information is displayed only if an actual libc function to get it is available. This also fixes warnings previously seen with musl libc: [CC] asterisk.c -> asterisk.o asterisk.c: In function 'handle_show_sysinfo': asterisk.c:773:6: warning: variable 'totalswap' set but not used [-Wunused-but-set-variable] int totalswap = 0; ^~~~~~~~~ asterisk.c:770:11: warning: variable 'freeswap' set but not used [-Wunused-but-set-variable] uint64_t freeswap = 0; ^~~~~~~~ Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca	9 years ago
zuul	95cf4f8d31	Merge "res_pjsip_transport_management: Convert time in log message to seconds."	9 years ago
zuul	544fe73811	Merge "chan_sip: Fix session timeout on retransmit of non-UDP packets"	9 years ago
zuul	f99155dd2e	Merge "rtp: Preserve timestamps on video frames."	9 years ago
zuul	e2d3882b30	Merge "sip_to_pjsip.py: Map legacy_useroption_parsing."	9 years ago
Joshua Colp	89764f7ae9	rtp: Preserve timestamps on video frames. Currently when receiving video over RTP we store only a calculated samples on the frame. When starting the video it can take some time for this calculation to actually yield a value as it requires constant changing timestamps. As well if a video frame passes over multiple RTP packets this calculation will fail as the timestamp is the same as the previous RTP packet and the number of samples calculated will be 0. This change preserves the timestamp on the frame and allows it to pass through the core. When sending the video this timestamp is used instead of a new one being calculated. ASTERISK-26367 #close Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd	9 years ago
zuul	cbd6f7001e	Merge "res_pjsip: Add ignore_uri_user_options option."	9 years ago
Joshua Colp	5f54ac3a80	res_pjsip_transport_management: Convert time in log message to seconds. ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc	9 years ago
zuul	87aa445559	Merge "res_pjsip: Don't assume a request will have any addresses."	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
zuul	c6aaf56be6	Merge "chan_sip: Allow target refresh (Contact update) on re-INVITE."	9 years ago
zuul	8076e78d50	Merge "res_pjsip_messaging.c: Misc cleanups and fixes."	9 years ago
Joshua Colp	e3487b9360	res_pjsip: Don't assume a request will have any addresses. When performing DNS resolution the failover code present in res_pjsip currently assumes that a request will always have at least one viable address. In practice this is not true. A domain may be used that has no records. The code now checks that at least one address exists on the request which prevents looping. ASTERISK-26364 #close Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c	9 years ago
Richard Mudgett	7d7b23f04f	app_queue: Fix CLI "queue show" and AMI Queues action output truncation. The output of CLI "queue show" and AMI Queues action is truncated and "failed to extend from 240 to 327" messages are generated if the queue member and interface names are lengthy. * Increase the string buffer size from 240 to 512 in order to accommodate for more information fields added to the output since v1.8. ASTERISK-26360 #close Reported by: Richard Mudgett Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d	9 years ago
zuul	50c3bb2631	Merge "contrib: Let safe_asterisk script continue without /dev/tty9."	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
Richard Mudgett	82ec58aa91	sip_to_pjsip.py: Map legacy_useroption_parsing. Map the sip.conf general section legacy_useroption_parsing to the new pjsip.conf global ignore_uri_user_options. ASTERISK-26316 Reported by: Kevin Harwell Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc	9 years ago

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