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2018-01-08res_pjsip: Add AMI action 'PJSIPShowAuths'Sungtae Kim
Add an AMI action which provides information on all configured Auths. ASTERISK-27547 Change-Id: I1a88a75b38a2b1dd9d1de6c0307b20a3f584c817
2018-01-08Merge "General: Avoid implicit conversion to char when changes value to ↵Jenkins2
negative."
2018-01-07res_stasis: Fix dial bridge unload.Corey Farrell
If the dial bridge has been created it must be released by calling ast_bridge_destroy, simply releasing the ao2 reference is not enough. Also move stasis_app_control_shutdown earlier in unload to ensure the bridge cannot be created or grabbed after the app_bridges container is released. Change-Id: I372302de94ca63876069e2585a049c5060e5e767
2018-01-07res_stasis: Fix app_is_subscribed_bridge_id.Corey Farrell
Instead of searching for bridge_id provided in an argument this function always searched for BRIDGE_ALL first. Rewrite this function to work like the similar functions for channel and endpoint functions. Change-Id: Ib5caca69e11727c5c8a7284a1d00621f40f1e60a
2018-01-06General: Silence modules on (un)load.Alexander Traud
Some (normally optional) modules created notices, warnings, and even errors in normal situations like (un)load. This cluttered the command-line interface (CLI) on start and while stopping gracefully. However, when an user went for the script './contrib/scripts/install_prereq', those modules get compiled-in because their prerequisites were met at compile time. Furthermore, because of ASTERISK_27475, the former talkative module 'res_curl' is built as side-effect. ASTERISK-27553 Change-Id: I9f105f46d72553994e820679bfde3478a551b281
2018-01-06General: Avoid implicit conversion to char when changes value to negative.Alexander Traud
clang 5.0 warned about this. ASTERISK-27557 Change-Id: I7cceaa88e147cbdf81a3a7beec5c1c20210fa41e
2018-01-05res_pjsip_endpoint_identifier_ip.c: Fix apply identify validation.Richard Mudgett
The ip_identify_apply() did not validate the configuration for simple static configuration errors or deal well with address resolution errors. * Added missing configuration validation checks. * Fixed address resolution error handling. * Demoted an error message to a warning since it does not fail applying the identify object configuration. Change-Id: I8b519607263fe88e8ce964f526a45359fd362b6e
2018-01-05res_pjsip.c: Fix endpoint identifier registration name search.Richard Mudgett
If an endpoint identifier name in the endpoint_identifier_order list is a prefix to the identifier we are registering, we could install it in the wrong position of the list. Assuming endpoint_identifier_order=username,ip,anonymous then registering the "ip_only" identifier would put the identifier in the wrong position of the priority list. * Fix incorrect strncmp() string prefix matching. Change-Id: Ib8819ec4b811da8a27419fd93528c54d34f01484
2018-01-04res_pjsip_history: Add missing unlock to CLI command.Corey Farrell
Change-Id: I872060a30543776a176a316309602d924a23eb29
2018-01-04Merge "loader: Create ast_module_running_ref."Joshua Colp
2018-01-04Merge "res_pjsip_session: Check if sequence header is missing"Joshua Colp
2018-01-03loader: Create ast_module_running_ref.Corey Farrell
This function returns NULL if the module in question is not running. I did not change ast_module_ref as most callers do not check the result and they always call ast_module_unref. Make use of this function when running registered items from: * app_stack API's * bridge technologies * CLI commands * File formats * Manager Actions * RTP engines * Sorcery Wizards * Timing Interfaces * Translators * AGI Commands * Fax Technologies ASTERISK-20346 #close Change-Id: Ia16fd28e188b2fc0b9d18b8a5d9cacc31df73fcc
2018-01-03Merge "res_pjsip: Add AMI action 'PJSIPShowAors'"Jenkins2
2018-01-03res_pjsip_session: Check if sequence header is missingKevin Harwell
The pjsip_msg_find_hdr function can return NULL. This patch adds a check when searching for the sequence header to make sure a NULL pointer is never de-referenced. Change-Id: I19af23aeeded65be016be92360e8cb7ffe51fad2
2018-01-02res_pjsip: Add AMI action 'PJSIPShowAors'Sungtae Kim
Add an AMI action which provides information on all configured AORs. ASTERISK-27537 Change-Id: If8b990a00909e5b6c0f04a3b8dccd9903dc445eb
2017-12-31ice: Increase foundation buffer sizeSean Bright
Per RFC 5245, the foundation specified with an ICE candidate can be up to 32 characters but we are only allowing for 31. ASTERISK-27498 #close Reported by: Michele Prà Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf
2017-12-22AST-2017-014: res_pjsip - Missing contact header can cause crashKevin Harwell
Those SIP messages that create dialogs require a contact header to be present. If the contact header was missing from the message it could cause Asterisk to crash. This patch checks to make sure SIP messages that create a dialog contain the contact header. If the message does not and it is required Asterisk now returns a "400 Missing Contact header" response. Also added NULL checks when retrieving the contact header that were missing as a "just in case". ASTERISK-27480 #close Change-Id: I1810db87683fc637a9e3e1384a746037fec20afe
2017-12-22Remove as much trailing whitespace as possible.Sean Bright
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
2017-12-20Fix Common Typo's.Corey Farrell
Fix instances of: * Retreive * Recieve * other then * different then * Repeated words ("the the", "an an", "and and", etc). * othterwise, teh ASTERISK-24198 #close Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
2017-12-20Merge "res_rtp_asterisk: Avoid close the rtp/rtcp fd twice."Jenkins2
2017-12-19Remove constant conditionals (dead-code).Corey Farrell
Some variables are set and never changed, making them constant. This means that code in the 'false' block of the conditional is unreachable. In chan_skinny and res_config_ldap I used preprocessor directive `#if 0` as I'm unsure if the unreachable code could be enabled in the future. Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059
2017-12-19res_rtp_asterisk: Avoid close the rtp/rtcp fd twice.Aaron An
When RTCP-MUX enabled. rtp->s is the same as rtcp->s, check this before close the file descriptor. Close the FD twice will hangs the asterisk under heavy load. ASTERISK-27299 #close Reported-by: Aaron An Tested-by: AaronAn Change-Id: I870a072d73fd207463ac116ef97100addbc0820a
2017-12-18Merge changes from topic 'faster-aco'Joshua Colp
* changes: aco: Minimize use of regex. aco: Create ways to minimize use of regex.
2017-12-15Merge "res_smdi: Fix shutdown ref."Jenkins2
2017-12-15Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams."Jenkins2
2017-12-15Merge "res_hep: hepv3_is_loaded() should check if we are enabled"Jenkins2
2017-12-15Merge "res_clialiases: Fix completion pass-through."Joshua Colp
2017-12-15Merge "coverity: Fix warnings in res_smdi"Jenkins2
2017-12-15Merge "res_musiconhold: Start playlist after initial announcement"Jenkins2
2017-12-15Merge "pjsip_options: wrongly applied "UNKNOWN" status"Jenkins2
2017-12-15aco: Minimize use of regex.Corey Farrell
Remove nearly all use of regex from ACO users. Still remaining: * app_confbridge has a legitamate use of option name regex. * ast_sorcery_object_fields_register is implemented with regex, all callers use simple prefix based regex. I haven't decided the best way to fix this in both 13/15 and master. Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
2017-12-15res_smdi: Fix shutdown ref.Corey Farrell
When adding shutdown refs for OPTIONAL_API components I accidentally added it to the unload_module function in res_smdi. Move it to load_module. Change-Id: I2b9da38fbc11ef78ea23dbb2df92b684be7f647c
2017-12-14res_hep: hepv3_is_loaded() should check if we are enabledSean Bright
res_hep_pjsip.so and res_hep_rtcp.so will still load and do a lot of unnecessary work even if 'enabled' is set to 'no' in hep.conf. Change-Id: I3eddfeea09c6b5bc7c641952ee0ae487fd09b64b
2017-12-14res_clialiases: Fix completion pass-through.Corey Farrell
Never ignore contents of line when generating completion options. Change-Id: I74389efdfea154019d3b56a9f381610614c044c8
2017-12-14Merge "res_pjsip_session: Reinvite using active stream topology if none ↵Jenkins2
requested."
2017-12-14res_rtp_asterisk.c: Disable packet flood detection for video streams.Richard Mudgett
We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70
2017-12-14res_pjsip_sdp_rtp: Add NULL check in add_crypto_to_streamGeorge Joseph
add_crypto_to_stream wasn't checking for a NULL session->inv_session->neg before calling pjmedia_sdp_neg_get_state. This was causing a crash if the negotiation hadn't already been completed and asterisk was compiled with --enable-dev-mode. Change-Id: I57c6229954a38145da9810fc18657bfcc4d9d0c9
2017-12-14res_musiconhold: Start playlist after initial announcementSean Bright
Reset the samples counter to zero when we are done playing an announcement so that we don't skip into the middle of the first file in the playlist. Also add the selected annoucement to the output of 'moh show classes.' ASTERISK-24329 #close Reported by: Thomas Frederiksen Change-Id: I2a5f986a31279c981592f49391409ebf38d6f6d0
2017-12-14coverity: Fix warnings in res_smdiSean Bright
ASTERISK-19657 #close Reported by: Matt Jordan III, Esq. Change-Id: I59a5e6ef3e7d9e848bec1f4b40cb73321bc7956a
2017-12-13Merge "pjsip_options: contacts sometimes not being updated on reload"Kevin Harwell
2017-12-13Merge "res_pjsip: Assign support levels to a few modules"Jenkins2
2017-12-13Merge "pjsip_options: dynamic contact's fields not updated on reload"Jenkins2
2017-12-13Merge "chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)"Joshua Colp
2017-12-13AST-2017-012: Place single RTCP report block at beginning of report.Joshua Colp
When the RTCP code was transitioned over to Stasis a code change was made to keep track of how many reports are present. This count controlled where report blocks were placed in the RTCP report. If a compound RTCP packet was received this logic would incorrectly place a report block in the wrong location resulting in a write to an invalid location. This change removes this counting logic and always places the report block at the first position. If in the future multiple reports are supported the logic can be extended but for now keeping a count serves no purpose. ASTERISK-27382 ASTERISK-27429 Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116
2017-12-13res_pjsip_session: Reinvite using active stream topology if none requested.Joshua Colp
When a connected line update is sent to an endpoint we do not request a specific stream topology to be used. Previously this resulted in the configured stream topology being used which may actually differ from the currently negotiated topology. PJSIP is helpful in this regard in that it will fill in any missing streams with removed ones. This results in our own state not matching the SDP, though, and we do not apply the negotiated SDP. This change tweaks the code to use the actively negotiated stream topology if it is present with a fallback to the configured one. This results in the SDP and the state having matching information and the world is happy. ASTERISK*27397 Change-Id: I7a57117f0183479e6884b7bf3a53bb8c7464f604
2017-12-12chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri)Richard Mudgett
This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33
2017-12-12res_pjsip: Add TLSv1.1 and TLSv1.2 supportSean Bright
Support for these protocols was added in the same commit as the 'proto' field, so we can safely use the same ./configure check. For reference: https://trac.pjsip.org/repos/changeset/4968 Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac
2017-12-12res_pjsip: Assign support levels to a few modulesSean Bright
Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d
2017-12-11pjsip_options: wrongly applied "UNKNOWN" statusKevin Harwell
A couple of places were setting the status to "UNKNOWN" when qualifies were being disabled. Instead this should be set to the "CREATED" status that represents when a contact is given (uri available), but the qualify frequency is set to zero so we don't know the status. This patch updates the relevant places with "CREATED". It also updates the "CREATED" status description (value shown in CLI/AMI/ARI output) to a value of "NonQualified"/"NonQual" as this description is hopefully less confusing. ASTERISK-27467 Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89
2017-12-11Merge "astdb: Improve prefix searches in astdb"Jenkins2
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