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2016-09-27codec_opus: Replace res_format_attr_opus with the one from codec_opusGeorge Joseph
Preparation ASTERISK-26409 Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3 (cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9)
2016-09-27format_ogg_opus: New formatGeorge Joseph
Add Ogg/Opus playback support. This uses libopusfile in order to be able to read .opus files and play them back. Writing/recording support is not present at this time. ASTERISK-26409 Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955
2016-09-25build_tools: Add ability to download variants to download_externalsGeorge Joseph
Some external packages have multiple variants that apply to different builds of asterisk. The DPMA for instance has a "bundled" variant that needs to be downloaded if asterisk was configured with --with-pjproject-bundled. There are 2 ways to specify variants: If you need the user to make the decision about which variant to download, simply create multiple menuselect "member" entries like so... <member name="res_digium_phone" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> </member> <member name="res_digium_phone-bundled" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> </member> Note that the second entry has "-<variant>" appended to the name. You can then use the existing menuselect facilities to restrict which members to enable or disable. Youy probably don't want the user to enable multiple at the same time. If you want to hide the details of the variants, the better way to do it is to create 1 member with "variant" elements. <member name="res_digium_phone" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> <member_data> <downloader> <variants> <variant tag="bundled" condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/> </variants> </downloader> </member_data> </member> The condition must be a bash expression suitable for use with an "if" statement. Any environment variable can be used plus those available in makeopts. In this case, if asterisk was configured with --with-pjproject-bundled the bundled variant will be automatically downloaded. Otherwise the normal version will be downloaded. Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e
2016-09-23Merge "channels/chan_pjsip: fix HANGUPCAUSE function bug." into 13zuul
2016-09-23Merge "chan_sip: Address runaway when realtime peers subscribe to ↵zuul
mailboxes" into 13
2016-09-23channels/chan_pjsip: fix HANGUPCAUSE function bug.Aaron An
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
2016-09-23chan_sip: Resolve externhost not to IPv6; instead go for IPv4.Alexander Traud
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
2016-09-23chan_sip: Address runaway when realtime peers subscribe to mailboxesGeorge Joseph
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
2016-09-22Merge "core: Ensure presencestate subtype and message are NULL." into 13zuul
2016-09-22Merge "res_odbc: Make pooling option deprecation notice more useful." into 13Joshua Colp
2016-09-21Merge "cdr_mysql: fix UTC support" into 13zuul
2016-09-21core: Ensure presencestate subtype and message are NULL.Joshua Colp
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: I61f8187972d5d8bbd7d6b7f4daa4f4f7e8237b23
2016-09-21Merge "logger: Fix default console settings." into 13zuul
2016-09-21res_odbc: Make pooling option deprecation notice more useful.Joshua Colp
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
2016-09-21Merge "core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get." into 13zuul
2016-09-21cdr_mysql: fix UTC supportTzafrir Cohen
* Make 'cdrzone=UTC' work properly. * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone ASTERISK-26359 #close Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778
2016-09-21odbc: Remove options that are no longer applicable.Joshua Colp
The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6
2016-09-21Merge "asterisk.c: Non-root users also get the astcanary after core ↵zuul
restart." into 13
2016-09-20core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get.Corey Farrell
Move the function outside the conditional block that excludes LOW_MEMORY. ASTERISK-26273 #close Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4
2016-09-20Merge "sd_notify (systemd status notifications) support" into 13Joshua Colp
2016-09-20Merge "res_pjsip_multihomed: Change Contact port to listening port." into 13zuul
2016-09-20logger: Fix default console settings.Corey Farrell
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
2016-09-20sd_notify (systemd status notifications) supportTzafrir Cohen
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 (cherry picked from commit 07b95f7c65b7c083724f1af2b26f93cc22cad58c)
2016-09-20asterisk.c: Non-root users also get the astcanary after core restart.Walter Doekes
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
2016-09-19Merge "asterisk.c: When astcanary dies on linux, reset priority on all ↵zuul
threads." into 13
2016-09-19Merge "Fix showing of swap details when sysinfo() is available" into 13zuul
2016-09-19Merge "res_config_odbc.c: Fix buffer size limitation creating invalid SQL." ↵zuul
into 13
2016-09-19asterisk.c: When astcanary dies on linux, reset priority on all threads.Walter Doekes
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
2016-09-16Fix showing of swap details when sysinfo() is availableTimo Teräs
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
2016-09-15res_config_odbc.c: Fix buffer size limitation creating invalid SQL.Richard Mudgett
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
2016-09-15res_pjsip_multihomed: Change Contact port to listening port.Joshua Colp
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
2016-09-15pjproject_bundled: Prevent SERVFAIL from marking name server badGeorge Joseph
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
2016-09-14Merge "res_pjsip_transport_management: Convert time in log message to ↵zuul
seconds." into 13
2016-09-14Merge "chan_sip: Fix session timeout on retransmit of non-UDP packets" into 13zuul
2016-09-14Merge "rtp: Preserve timestamps on video frames." into 13zuul
2016-09-14Merge "sip_to_pjsip.py: Map legacy_useroption_parsing." into 13zuul
2016-09-14rtp: Preserve timestamps on video frames.Joshua Colp
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
2016-09-14Merge "res_pjsip: Add ignore_uri_user_options option." into 13zuul
2016-09-14res_pjsip_transport_management: Convert time in log message to seconds.Joshua Colp
ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc
2016-09-13chan_sip: Fix session timeout on retransmit of non-UDP packetsSteve Davies
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
2016-09-13Merge "chan_sip: Allow target refresh (Contact update) on re-INVITE." into 13zuul
2016-09-13Merge "res_pjsip_messaging.c: Misc cleanups and fixes." into 13zuul
2016-09-12app_queue: Fix CLI "queue show" and AMI Queues action output truncation.Richard Mudgett
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
2016-09-12Merge "contrib: Let safe_asterisk script continue without /dev/tty9." into 13zuul
2016-09-12chan_sip: Allow target refresh (Contact update) on re-INVITE.Walter Doekes
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
2016-09-09sip_to_pjsip.py: Map legacy_useroption_parsing.Richard Mudgett
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
2016-09-09res_pjsip: Add ignore_uri_user_options option.Richard Mudgett
This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
2016-09-09contrib: Let safe_asterisk script continue without /dev/tty9.Walter Doekes
If you use the safe_asterisk script, it uses hardcoded defaults before running configurable values from /etc/asterisk/startup.d. The hardcoded default has TTY=9. Some containerized environments don't have such a TTY, and safe_asterisk would stop. The custom configuration from /etc/asterisk/startup.d/* isn't read until after it stopped, so changing TTY in a custom config did not help. This changeset changes safe_asterisk to continue if the TTY setting was untouched and /dev/tty9 and /dev/vc/9 aren't found. Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc
2016-09-09res_pjsip: Only invoke unidentified endpoint logic when unidentified.Joshua Colp
The code was incorrectly invoking the unidentified logic when an endpoint had actually been identified, causing log messages to be output. ASTERISK-26349 #close Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f
2016-09-09chan_sip: Don't allocate new RTP instances on top of old ones.Joshua Colp
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
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