32748 Commits (fdc13060df8f79cd5277fb2c8a6c70d22b5cb69c)

 

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Author	SHA1	Message	Date
Richard Mudgett	fe3cce816c	app_chanisavail.c: Simplify dialplan using ChanIsAvail. ... Dialplan has to be careful about passing an empty device list or empty positions in the list. As a result, dialplan has to check for these conditions before using ChanIsAvail. Simplify dialplan by making ChanIsAvail handle these conditions gracefully. * Made tolerate empty positions in the device list. * Simplified the code and eliminated some unnecessary indention. ASTERISK-28638 Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3	6 years ago
George Joseph	1c9ddad4db	stasis.c: Use correct topic name in stasis_topic_pool_delete_topic ... When a topic is created for an object, its name is only <object>:<uniqueid> For example: bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 When a topic is added to a pool, its name has the pool's topic name prepended. For example: bridge:all/bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 The topic_pool_entry's name however, is only what was passed in to stasis_topic_pool_get_topic which is bridge:cb68b3a8-fce7-4738-8a17-d7847562f020 That's actually correct because the entry is qualified by the pool that's in. When you're ready to delete the entry from the pool, you retrieve the tropic name from the object but since it now has the pool's topic name prepended, it won't be found in the pool container. Fix: * Modified stasis_topic_pool_delete_topic() to skip past the pool topic's name, if it was prepended to the topic name, before searching the container for a pool entry. ASTERISK-28633 Reported by: Joeran Vinzens Change-Id: I4396aa69dd83e4ab84c5b91b39293cfdbcf483e6	6 years ago
Richard Mudgett	19069f7db7	app_bridgeaddchan.c: Make BridgeAdd be more like Bridge ... * Made BridgeAdd not hangup the call if there is a problem. * Reduced message level from warning to verbose for normal exception cases. * Added a loop safety check to BridgeAdd. * Made BridgeAdd set BRIDGERESULT with the status when dialplan is resumed. Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426	6 years ago
Richard Mudgett	abcb4ab321	app_dial.c: Simplify dialplan using Dial. ... Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Dial. Simplify dialplan by making Dial handle these conditions gracefully. * Made tolerate empty positions in the dialed device list. * Reduced some message log levels from notice to verbose. ASTERISK-28638 Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9	6 years ago
Richard Mudgett	d86a6ac5ce	app_page.c: Simplify dialplan using Page. ... Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Page. Simplify dialplan by making Page handle these conditions gracefully. * Made tolerate empty positions in the paged device list. * Reduced some warnings associated with the 's' option to verbose messages. The warning level for those messages really serves no purpose as that is why the 's' option exists. ASTERISK-28638 Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3	6 years ago
Richard Mudgett	0376f2bba9	features.c: Make Bridge application tolerate unspecified channel. ... The Bridge application was inconsistent if the channel to bridge with is not specified. If no parameters are given then a warning is issued and the current channel is hung up. If options are given but no channel is specified then a warning is issued and the current channel is not hung up. * Made the Bridge application give a verbose message instead of a warning if the channel to bridge with is not specified and made not hang up the current channel. As a result dialplan no longer needs to check if a channel name is passed before calling Bridge and simply needs to check the BRIDGERESULT channel variable instead. This is something you likely want your dialplan to do anyway. * Fixed up L() option warning message. It is up to the caller to determine if the channel is hung up because of the warning. Dial() hangs up the current channel while Bridge() does not. Change-Id: I44349a8dc3912397f28852777de04f19e7bb9c73	6 years ago
Richard Mudgett	0d1f3d9bf3	app_chanspy.c: Reduce log message level from notice to verbose. ... Change-Id: Ica5f38ccd8cdc077aef14d0c50425e0b29ac7e0a	6 years ago
Richard Mudgett	a457947198	app_softhangup.c: Reduce unnecessary warning to verbose message. ... Why log a warning for something your dialplan explicitly asked for? Change-Id: I167b90daf4c7d75dd4b7ef94849f6cef05aa43a7	6 years ago
Sean Bright	b40dd11afe	res_pjsip_config_wizard: Fix change detection for wizard settings ... ast_sorcery_changeset_create() is not commutative and will fail to detect differences between two variable lists depending on what changed, so switch to ast_variable_lists_match(). ASTERISK-28492 #close Reported by: Jean-Denis Girard Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1	6 years ago
Sean Bright	7d94bdde9d	res_agi: Improve GET FULL VARIABLE documentation ... ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f	6 years ago
Sean Bright	87110c1bdf	websocket: Consider pending SSL data when waiting for socket input ... When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d	6 years ago
Joshua Colp	cee68ea4f2	Merge "func_odbc: acf_odbc_read() and cli_odbc_read() unicode support"	6 years ago
Jean Aunis	034ac357ad	ARI: Ability to inhibit COLP frames when adding channels to a bridge ... This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel' operation in the Bridges REST API. When set, this flag avoids generating COLP frames when the specified channels enter the bridge. ASTERISK-28629 Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc	6 years ago
George Joseph	be93537382	Merge "res_fax: wrap v21 detected Asterisk initiated negotiation with config option"	6 years ago
Friendly Automation	2a8f759374	Merge "chan_sip: voice frames are no longer transmitted after emitting a COLP"	6 years ago
Sean Bright	fc99ac8c9a	db: Initialize condition primitive before use ... The db_init() function ultimately calls db_sync() which signals the condition before it is initialized. Change-Id: Id4a4e025b637bc4ac7d90557fcb71d56598892ab	6 years ago
George Joseph	7c4fc2e39a	Merge "config.c: Skip UTF-8 BOMs if present when reading config files"	6 years ago
Joshua C. Colp	dd823e50e4	Merge "main/file.c: Limit media cache usage to remote files."	6 years ago
Joshua C. Colp	d21427cadd	Merge "app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR."	6 years ago
Friendly Automation	c3cf0e330c	Merge "chan_sip: in case of tcp/tls, be less annoying about tx errors."	6 years ago
Friendly Automation	07b7c6f50a	Merge "confbridge: Add support for specifying maximum sample rate."	6 years ago
Sean Bright	40b5cf8f52	config.c: Skip UTF-8 BOMs if present when reading config files ... ASTERISK-28667 #close Change-Id: I4767ed365c98f3e1587b7653321048a31d8a53b2	6 years ago
Kevin Reeves	c626ccec12	main/file.c: Limit media cache usage to remote files. ... When testing for the existance of a file, the media cache is searched even if the file has no chance of being in it. This can cause performance issues as the media cache size increases. As a result, calls to applications like Read and Playback using local files must scan through the media cache before playing. Under load and with a large cache, this can delay the playback of those files. This patch updates the function that checks for the existance of a file to only consult the media cache database if the requested file is a remote path. It introduces a new is_remote_path() function in main/file.c. ASTERISK-28625 #close Reported-by: kevin@phoneburner.com Change-Id: If91137493732d9034dafa381c081c69274a7dcc9	6 years ago
George Joseph	657ada8bfd	Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read"	6 years ago
Friendly Automation	308592a681	Merge "sip_to_pjsip.py: Fix trustrpid typo"	6 years ago
Joshua C. Colp	c29b7c00e5	Merge "configure: Add check for MySQL client bool and my_bool type usage."	6 years ago
Joshua C. Colp	2779189689	Merge "res_pjsip_session: Set stream state on created streams for incoming SDP."	6 years ago
Snuffy	095c204fe0	contrib/valgrind: Fix use of frame-level suppression ... Fix use of frame-level wildcard usage in suppression file. ASTERISK-27243 #close Reported-by: Richard Kenner Change-Id: I1c0c64c5f305d2c9aa124e11f1f64a2eec52dc51	6 years ago
Pascal Cadotte Michaud	e494d5fd76	sip_to_pjsip.py: Fix trustrpid typo ... ASTERISK-28664 #close Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6	6 years ago
Frederic LE FOLL	a83625b366	app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR. ... Temporary channel lifespan is very short and CDR deactivation request through ast_cdr_set_property() may happen when CDR is not available yet. Use CDR_PROP() dialplan function instead, it will first wait for pending CDR insertion requests to be processed. ASTERISK-28636 Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074	6 years ago
George Joseph	8587ef3a0c	Merge "res_pjsip_nat: Restore original contact for REGISTER responses"	6 years ago
Joshua C. Colp	ed394ce5b1	configure: Add check for MySQL client bool and my_bool type usage. ... Instead of trying to use the defined MySQL client version from the header use a configure check to determine whether the bool or my_bool type should be used for defining a boolean. ASTERISK-28604 Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7	6 years ago
Joshua C. Colp	89b7144fbd	confbridge: Add support for specifying maximum sample rate. ... ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	6 years ago
Friendly Automation	7e441d6b25	Merge "PJSIP_CONTACT: add missing argument documentation"	6 years ago
Joshua C. Colp	a603d7d324	res_pjsip_session: Set stream state on created streams for incoming SDP. ... A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355	6 years ago
Kevin Harwell	b6f5607359	res_fax: wrap v21 detected Asterisk initiated negotiation with config option ... A previous patch: Gerrit Change-Id: I73bb24799bfe1a48adae9c034a2edbae54cc2a39 made it so a T.38 Gateway tries to negotiate with both sides by sending T.38 negotiation request to both endpoints supported T.38 versus the previous behavior of forwarding negotiation to the "other" channel once a preamble was detected. This had the unfortunate side effect of breaking some setups. Specifically ones that set the max datagram option on an endpoint configuration (configured max datagram was not propagated since Asterisk now initiates negotiations). This patch adds a configuration option, "negotiate_both", that when enabled makes it so Asterisk initiates the negotiation requests to both endpoints vs. the previous behavior of waiting, and forwarding the request. The default is disabled keeping with the old behavior. ASTERISK-28660 Change-Id: I5deb875f3485e20bc75119ec743090655d864a1a	6 years ago
Jaco Kroon	32160cb456	ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. ... Due to use in res_rtp_asterisk there is a need to be able to apply an ACL without logging any invalid/denies. It's probably sensible to at least validate the ACL once directly after load and report invalid ACLs. Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Pascal Cadotte Michaud	bf4dd3d837	PJSIP_CONTACT: add missing argument documentation ... add missing argument "rtt" and "status" to the documentation The change to the dtd file allow an enumlist to contain one or many configOptionToEnum or enum. This is different from the previous patch I submitted when you could have a configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or (one or many enums) ASTERISK-28626 Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d	6 years ago
George Joseph	91cdb9537e	Merge "Revert "PJSIP_CONTACT: add missing argument documentation""	6 years ago
Joshua Colp	d0b198b330	Revert "PJSIP_CONTACT: add missing argument documentation" ... This reverts commit 7e3015d779. Reason for revert: Regression in XML validation. validity error : Content model of enumlist is not determinist: (configOptionToEnum | (configOptionToEnum , enum+) | enum+) As we are preparing to do releases and this is not critical I am reverting this for now until resolved. Change-Id: I30c2295f9d7f0a0475674ee77071a7ebabf5b83f	6 years ago
George Joseph	39c920ac78	res_rtp_asterisk: Add frame list cleanups to ast_rtp_read ... In Asterisk 16+, there are a few places in ast_rtp_read where we've allocated a frame list but return a null frame instead of the list. In these cases, any frames left in the list won't be freed. In the vast majority of the cases, the list is empty when we return so there's nothing to free but there have been leaks reported in the wild that can be traced back to frames left in the list before returning. The escape paths now all have logic to free frames left in the list. ASTERISK-28609 Reported by: Ted G Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a	6 years ago
George Joseph	d8dac784f4	Merge "res_pjsip_registrar.c: Prevent potential double free if AOR is not found"	6 years ago
Jaco Kroon	365d007eb6	chan_sip: in case of tcp/tls, be less annoying about tx errors. ... chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to 213.150.203.60:1492 returned -2: Interrupted system call returned -2 implies this wasn't actually an OS error, so errno makes no sense either. Internal error was already logged higher up, and -2 generally means that either there isn't a valid connection available, or the pipe notification failed, and that is already correctly logged. ASTERISK-28651 #close Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Friendly Automation	616f702865	Merge "app_queue: Fix old confusing comment about when the members are called"	6 years ago
George Joseph	cbc1136704	res_pjsip_nat: Restore original contact for REGISTER responses ... RFC3261 Section 10 "Registrations", specifically paragraph "10.2.4: Refreshing Bindings", states that a user agent compares each contact address (in a 200 REGISTER response) to see if it created the contact. If the Asterisk endpoint has the rewrite_contact option set however, the contact host and port sent back in the 200 response will be the rewritten one and not the one sent by the user agent. This prevents the user agent from matching its own contact. Some user agents get very upset when this happens and will not consider the registration successful. While this is rare, it is acceptable behavior especially if more than 1 user agent is allowed to register to a single endpoint/aor. This commit updates res_pjsip_nat (where rewrite_contact is implemented) to store the original incoming Contact header in a new "x-ast-orig-host" URI parameter before rewriting it, and to restore the original host and port to the Contact headers in the outgoing response. This is only done if the request is a REGISTER and rewrite_contact is enabled. pjsip_message_filter was also updated to ensure that if a request comes in with any existing x-ast-* URI parameters, we remove them so they don't conflict. Asterisk will never send a request with those headers in it but someone might just decide to add them to a request they craft and send to Asterisk. NOTE: If a device changes its contact address and registers again, it's a NEW registration. If the device didn't unregister the original registration then all existing behavior based on aor/remove_existing and aor/max_contacts apply. ASTERISK-28502 Reported-by: Ross Beer Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e	6 years ago
Friendly Automation	ec559c34db	Merge "res_pjsip_outbound_registration: add support for SRV failover"	6 years ago
Friendly Automation	e99cdeeff3	Merge "res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases"	6 years ago
Friendly Automation	fd00a2bd10	Merge "channel.c: Resolve issue with receiving SIP INFO packets for DTMF"	6 years ago
Joshua Colp	702581647f	Merge "chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime."	6 years ago
Friendly Automation	2587b7e45f	Merge "PJSIP_CONTACT: add missing argument documentation"	6 years ago