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2017-09-15Merge "res_pjsip: Filter out non SIP(S) requests" into 13Jenkins2
2017-09-15Merge "res_calendar: Various fixes" into 13Jenkins2
2017-09-14res_pjsip: Filter out non SIP(S) requestsGeorge Joseph
Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460
2017-09-14Merge "chan_rtp: Use μ-law by default instead of signed linear" into 13Jenkins2
2017-09-14Merge "res_pjsip: Add handling for incoming unsolicited MWI NOTIFY" into 13Jenkins2
2017-09-13res_calendar: Various fixesSean Bright
* The way that we were looking at XML elements for CalDAV was extremely fragile, so use SAX2 for increased robustness. * Don't complain about a 'channel' not be specified if autoreminder is not set. Assume that if 'channel' is not set, we don't want to be notified. * Fix some truncated CLI output in 'calendar show calendar' and make the 'Autoreminder' description a bit more clear ASTERISK-24588 #close Reported by: Stefan Gofferje ASTERISK-25523 #close Reported by: Jesper Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c
2017-09-13chan_rtp: Use μ-law by default instead of signed linearSean Bright
Multicast/Unicast RTP do not use SDP so we need to use a format that cleanly maps to one of the static RTP payload types. Without this change, an Originate to a Multicast or Unicast channel without a format specified would produce no audio on the receiving device. ASTERISK-21399 #close Reported by: Tzafrir Cohen Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3
2017-09-13res_pjsip: Add handling for incoming unsolicited MWI NOTIFYGeorge Joseph
A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c
2017-09-12Merge "alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode" into 13Jenkins2
2017-09-11res_rtp_asterisk.c: Add doxygen to RTCP payload types.Richard Mudgett
Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998
2017-09-11Merge "alembic: Add support for MS-SQL" into 13Jenkins2
2017-09-11alembic: Fix typo in add_auto_info_to_endpoint_dtmf_modeGeorge Joseph
The downgrade function was missing "_v2" at the end of the alter column type. Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e
2017-09-10res/res_pjsip: Fix localnet checks in pjsip, part 2.Walter Doekes
In 45744fc53, I mistakenly broke SDP media address rewriting by misinterpreting which address was checked in the localnet comparison. Instead of checking the remote peer address to decide whether we need media address rewriting, we check our local media address: if it's local, then we rewrite. This feels awkward, but works and even made directmedia work properly if you set local_net. (For the record: for local peers, the SDP media rewrite code is not called, so the comparison does no harm there.) ASTERISK-27248 #close Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f
2017-09-08alembic: Add support for MS-SQLFlorian Floimair
MS-SQL has no native Enum-type support and therefore needs to work with constraints. Since these constraints need unique names the suggested approach referenced in the following alembic documentation has been applied: http://bit.ly/2x9r8pb ASTERISK-27255 #close Change-Id: I4a399ba3eed41a33ce8cb294968ad340221580ee
2017-09-08Merge "app_waitforsilence: Cleanup & don't treat missing frames as 'noise'" ↵Jenkins2
into 13
2017-09-08Merge "chan_sip: when getting sip pvt return failure if not found" into 13Joshua Colp
2017-09-07Merge "res_srtp: Add support for libsrtp2.1." into 13Jenkins2
2017-09-07Merge "chan_sip: Do not change IP address in SDP origin line (o=) in SIP ↵Jenkins2
reINVITE" into 13
2017-09-07Merge "func_cdr: honour 'u' flag on dummy channel" into 13Jenkins2
2017-09-07func_cdr: honour 'u' flag on dummy channelJacek Konieczny
Fixes ${CDR(...,u)} when used in cdr_custom.conf ASTERISK-27165 #close Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83
2017-09-06chan_sip: when getting sip pvt return failure if not foundScott Griepentrog
In handle_request_invite, when processing a pickup, a call is made to get_sip_pvt_from_replaces to locate the pvt for the subscription. The pvt is assumed to be valid when zero is returned indicating no error, and is dereferenced which can cause a crash if it was not found. This change checks the not found case and returns -1 which allows the calling code to fail appropriately. ASTERISK-27217 #close Reported-by: Bryan Walters Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612
2017-09-06app_waitforsilence: Cleanup & don't treat missing frames as 'noise'Sean Bright
* WaitForSilence completes successfully if it receives no media in the specified timeout, but when acting as WaitForNoise that logic needs to be reversed. * Use standard argument parsing macros and add some error checking for invalid values. * The documentation indicated that the first argument to both WaitForSilence and WaitForNoise was required when it was not. Update the documentation to reflect that. * Wrap up some behavior in structs to avoid boolean checks all over the place. ASTERISK-24066 #close Reported by: M vd S Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9
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 13Joshua Colp
2017-09-06Merge "alembic: fix erroneous commit for add_prune_on_boot" into 13Jenkins2
2017-09-06Merge "res/res_pjsip: Standardize/fix localnet checks across pjsip." into 13Jenkins2
2017-09-06Merge "res_rtp_asterisk.c: Check RTP packet version earlier." into 13Jenkins2
2017-09-06chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITEVitezslav Novy
If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE to both parties to set up media path directly between the endpoints. In this reINVITE msg SDP origin line (o=) contains IP address of endpoint instead of IP of asterisk. This behavior violates RFC3264, sec 8: "When issuing an offer that modifies the session, the "o=" line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP." This patch assures IP address of Asterisk is always sent in SDP origin line. ASTERISK-17540 Reported by: saghul Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e
2017-09-06Merge "formats: Restore previous fread() behavior" into 13Jenkins2
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 13
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
2017-09-06res_srtp: Add support for libsrtp2.1.Alexander Traud
Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with previous versions of libSRTP, this change allows libSRTP 2.1.x as well. ASTERISK-27253 #close Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315
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
2017-09-05res_rtp_asterisk.c: Check RTP packet version earlier.Richard Mudgett
Change-Id: Ic6493a7d79683f3e5845dff1cee49445fd5a0adf
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 13Joshua Colp
2017-09-05Merge "chan_ooh323: Fix confusing indentation warning" into 13Jenkins2
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 13Jenkins2
2017-08-31Merge "pjsip_message_ip_updater: Fix issue handling "tel" URIs" into 13Jenkins2
2017-08-31Merge "AST-2017-006: Fix app_minivm application MinivmNotify command ↵Jenkins2
injection" into 13
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
2017-08-30Merge "bridge_native_rtp.c: Fixup native_rtp_framehook()" into 13Jenkins2
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