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2017-09-06stasis/control: Fix possible deadlock with swap channelGeorge Joseph
If an error occurs during a bridge impart it's possible that the "bridge_after" callback might try to run before control_swap_channel_in_bridge has been signalled to continue. Since control_swap_channel_in_bridge is holding the control lock and the callback needs it, a deadlock will occur. * control_swap_channel_in_bridge now only holds the control lock while it's actually modifying the control structure and releases it while the bridge impart is running. * bridge_after_cb is now tolerant of impart failures. Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3
2017-09-06Merge "alembic: Fix enum creation for dtls_fingerprint" into 15George Joseph
2017-09-06Merge "alembic: fix erroneous commit for add_prune_on_boot" into 15Jenkins2
2017-09-06Merge "res/res_pjsip: Standardize/fix localnet checks across pjsip." into 15Joshua Colp
2017-09-06Merge "formats: Restore previous fread() behavior" into 15Jenkins2
2017-09-06alembic: Fix enum creation for dtls_fingerprintGeorge Joseph
Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db
2017-09-06Merge "res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL ↵Jenkins2
channel" into 15
2017-09-06Merge "res_calendar*, res_smdi: Move to "extended" support" into 15Joshua Colp
2017-09-06alembic: fix erroneous commit for add_prune_on_bootFlorian Floimair
Added include for postgresql ENUM type and redefined values in the same way as in the other migration scripts. ASTERISK-27254 #close Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa
2017-09-05chan_pjsip: Suppress frame warnings.Ben Ford
When rtp_keepalive is on for a PJSIP endpoint dialing to another Asterisk instance also using PJSIP, Asterisk will continue to print warning messages about not being able to send frames of a certain type. This suppresses that warning message. Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67
2017-09-05formats: Restore previous fread() behaviorSean Bright
Some formats are able to handle short reads while others are not, so restore the previous behavior for the format modules so that we don't have spurious errors when playing back files. ASTERISK-27232 #close Reported by: Jens T. Change-Id: Iab7f52b25a394f277566c8a2a4b15a692280a300
2017-09-05res/res_pjsip: Standardize/fix localnet checks across pjsip.Walter Doekes
In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was confusion about whether the transport_state->localnet ACL has ALLOW or DENY semantics. For the record: the localnet has DENY semantics, meaning that "not in the list" means ALLOW, and the local nets are in the list. Therefore, checks like this look wrong, but are right: /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */ if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) { ast_debug(5, "Request is being sent to local address, " "skipping NAT manipulation\n"); (In the list == localnet == DENY == skip NAT manipulation.) And conversely, other checks that looked right, were wrong. This change adds two macro's to reduce the confusion and uses those instead: ast_sip_transport_is_nonlocal(transport_state, addr) ast_sip_transport_is_local(transport_state, addr) ASTERISK-27248 #close Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934
2017-09-05Merge "app_directory: Handle a NULL mailbox without crashing" into 15Jenkins2
2017-09-05res_calendar*, res_smdi: Move to "extended" supportGeorge Joseph
Change-Id: I31eee8be30c6b0fc3dadb31111dd47742da8892d
2017-09-05Merge "chan_ooh323: Fix confusing indentation warning" into 15Jenkins2
2017-09-05res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channelGeorge Joseph
t38_reinvite_response_cb can get called by res_pjsip_session's session_inv_on_tsx_state_changed in situations where session->channel is NULL. If it is, the ast_log warning segfaults because it tries to get the channel name from a NULL channel. * Check session->channel and print "unknown channel" when it's NULL. ASTERISK-27236 Reported by: Ross Beer Change-Id: I4326e288d36327f6c79ab52226d54905cdc87dc7
2017-09-01rtp_engine: Prevent possible double free with DTLS configSean Bright
ASTERISK-27225 #close Reported by: Richard Kenner Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89
2017-09-01chan_ooh323: Fix confusing indentation warningSean Bright
ASTERISK-27177 #close Reported by: Tzafrir Cohen Change-Id: I40311c404edb2302a7543ad5ca7a06b2a38f2d97
2017-09-01app_directory: Handle a NULL mailbox without crashingSean Bright
ASTERISK-27241 #close Reported by: David Moore Change-Id: Ibbbca85517b04c315406ebfe3b6f7e0763daedc6
2017-08-31Merge "chan_pjsip: Add tag info in CHANNEL function" into 15Jenkins2
2017-08-31Merge "res_rtp_asterisk: Allow remote SSRC to change on an RTP instance." ↵Jenkins2
into 15
2017-08-31Merge "AST-2017-006: Fix app_minivm application MinivmNotify command ↵Joshua Colp
injection" into 15
2017-08-31Merge "res_rtp_asterisk: Only learn a new source in learn state." into 15Joshua Colp
2017-08-31Merge "pjsip_message_ip_updater: Fix issue handling "tel" URIs" into 15Joshua Colp
2017-08-30Merge "confbridge: Handle user hangup during name recording" into 15Joshua Colp
2017-08-30pjsip_message_ip_updater: Fix issue handling "tel" URIsGeorge Joseph
sanitize_tdata was assuming all URIs were SIP URIs so when a non SIP uri was in the From, To or Contact headers, the unconditional cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused a segfault when trying to access uri->other_param. * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri) checks before attempting to cast or use the returned uri. ASTERISK-27152 Reported-by: Ross Beer Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f
2017-08-30AST-2017-006: Fix app_minivm application MinivmNotify command injectionCorey Farrell
An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
2017-08-30res_rtp_asterisk: Only learn a new source in learn state.Joshua Colp
This change moves the logic which learns a new source address for RTP so it only occurs in the learning state. The learning state is entered on initial allocation of RTP or if we are told that the remote address for the media has changed. While in the learning state if we continue to receive media from the original source we restart the learning process. It is only once we receive a sufficient number of RTP packets from the new source that we will switch to it. Once this is done the closed state is entered where all packets that do not originate from the expected source are dropped. The learning process has also been improved to take into account the time between received packets so a flood of them while in the learning state does not cause media to be switched. Finally RTCP now drops packets which are not for the learned SSRC if strict RTP is enabled. ASTERISK-27013 Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c
2017-08-30res_rtp_asterisk: Allow remote SSRC to change on an RTP instance.Joshua Colp
When SDP renegotiation occurs it is possible for an RTP instance to be reused for a new stream, resulting in the remote SSRC changing if it is part of a bundle group. This change allows this and updates its mapping in the current bundle group. ASTERISK-27231 Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490
2017-08-30chan_pjsip: Add tag info in CHANNEL functionAndre Nazario
Create local_tag and remote_tag in CHANNEL info to get tag from From and To headers of a SIP dialog. ASTERISK-27220 Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524
2017-08-29bridge_native_rtp.c: Fixup native_rtp_framehook()Richard Mudgett
* Fix framehook to test frame type for control frame. * Made framehook exit early if frame type is not a control frame. * Eliminated RAII_VAR in framehook. * Use switch instead of else-if ladder for control frame handling. Change-Id: Ia555fc3600bd85470e3c0141147dbe3ad07c1d18
2017-08-29confbridge: Handle user hangup during name recordingSean Bright
This prevents orphaned CBAnn channels from getting stuck in the bridge. ASTERISK-26994 #close Reported by: James Terhune Change-Id: I5e43e832a9507ec3f2c59752cd900b41dab80457
2017-08-29Merge "core: Reduce video update queueing." into 15Jenkins2
2017-08-29Merge "app_record: Resolve some absolute vs. relative filename bugs" into 15Joshua Colp
2017-08-29Merge "voicemail: Fix various abuses of mkstemp" into 15Jenkins2
2017-08-28core: Reduce video update queueing.Joshua Colp
A video update frame is used to indicate that a channel with video negotiated should provide a full frame so the decoder decoding the stream is able to do so. In situations where a queue is used to store frames it makes no sense for the queue to contain multiple video update frames. One is sufficient to have a full frame be sent. ASTERISK-27222 Change-Id: Id3f40a6f51b740ae4704003a1800185c0c658ee7
2017-08-28Merge "res/res_pjsip_session: allow SDP answer to be regenerated" into 15Jenkins2
2017-08-28Merge "alembic: Add dtls_fingerprint column in ps_endpoints table" into 15Joshua Colp
2017-08-25voicemail: Fix various abuses of mkstempSean Bright
mkstemp() returns a unique filename, but appending an extension to that filename does not guarantee uniqueness. Instead, use mkdtemp() and we can put whatever extension we want on the files that we create inside the directory. In the case of app_minivm, we also now properly clean up any temporary files that we create. ASTERISK-20858 #close Reported by: Walter Doekes Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43
2017-08-25app_record: Resolve some absolute vs. relative filename bugsSean Bright
If the Record() application is called with a relative filename that includes directories, we were not properly creating the intermediate directories and Record() would fail. Secondarily, updated the documentation for RECORDED_FILE to mention that it does not include a filename extension. Finally, rewrote the '%d' functionality to be a bit more straight forward and less noisy. ASTERISK-16777 #close Reported by: klaus3000 Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2
2017-08-25Merge "app_queue: Evaluate realtime queues when running dialplan functions" ↵Jenkins2
into 15
2017-08-25Merge "chan_pjsip.c: Fix topology refresh response code accuracy." into 15Joshua Colp
2017-08-25Merge "app_voicemail: Honor escape digits in "greeting only" mode" into 15Jenkins2
2017-08-25alembic: Add dtls_fingerprint column in ps_endpoints tableFlorian Floimair
The ps_endpoints table was missing the dtls_fingerprint column introduced with commit adba2a8d7fd. ASTERISK-27168 #close Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd
2017-08-25res/res_pjsip_session: allow SDP answer to be regeneratedTorrey Searle
If an SDP answer hasn't been sent yet, it's legal to change it. This is required for PJSIP_DTMF_MODE to work correctly, and can also have use in the future for updating codecs too. ASTERISK-27209 #close Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1
2017-08-24app_queue: Evaluate realtime queues when running dialplan functionsSean Bright
ASTERISK-19103 #close Reported by: Jim Van Meggelen Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b
2017-08-24app_voicemail: Honor escape digits in "greeting only" modeSean Bright
ASTERISK-21241 #close Reported by: Eelco Brolman Patches: Patch uploaded by Eelco Brolman (License 6442) Change-Id: Icbe39b5c82a49b46cf1d168dc17766f3d84f54fe
2017-08-24res_smdi: Clean up memory leakSean Bright
Change-Id: I1e33290929e1aa7c5b9cb513f8254f2884974de8
2017-08-23Merge "res_pjsip_session.c: Fix crash when declining an active stream." into 15Joshua Colp
2017-08-23Merge changes from topic 'ASTERISK-27212' into 15Jenkins2
* changes: bridge_channel.c: Fix FRACK when mapping frames to the bridge. bridge: Fix softmix bridge deadlock.
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