4161 Commits (ee8ea9275ffb0adf20faae2e8fce4bbad330e994)

Author	SHA1	Message	Date
Ben Ford	1274117102	res_stir_shaken: Add outbound INVITE support. ... Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is sent, the caller ID will be checked to see if there is a certificate that corresponds to it. If so, that information will be retrieved and an Identity header will be added to the SIP message. The format is: header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken Header, payload, and signature are all BASE64 encoded. The public key URL is retrieved from the certificate. Currently the algorithm and ppt are ES256 and shaken, respectively. This message is signed and can be used for verification on the receiving end. Two new configuration options have been added to the certificate object: attestation and origid. The attestation is required and must be A, B, or C. origid is the origination identifier. A new utility function has been added as well that takes a string, allocates space, BASE64 encodes it, then returns it, eliminating the need to calculate the size yourself. Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4	6 years ago
Kevin Harwell	415b55af5a	pjproject: Upgrade bundled version to pjproject 2.10 ... This patch makes the usual necessary changes when upgrading to a new version pjproject. For instance, version number bump, patches removed from third-party, new *.md5 file added, etc.. This patch also includes a change to the Asterisk pjproject Makefile to explicitly create the 'source/pjsip-apps/lib' directory. This directory is no longer there by default so needs to be added so the Asterisk malloc debug can be built. This patch also includes some minor changes to Asterisk that were a result of the upgrade. Specifically, there was a backward incompatibility change made in 2.10 that modified the "expires header" variable field from a signed to an unsigned value. This potentially effects comparison. Namely, those check for a value less than zero. This patch modified a few locations in the Asterisk code that may have been affected. Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to check a minimum version of pjproject at compile time. ASTERISK-28899 #close Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81	6 years ago
Joshua C. Colp	de2813cf23	core_unreal / core_local: Add multistream and re-negotiation. ... When requesting a Local channel the requested stream topology or a converted stream topology will now be placed onto the resulting channels. Frames written in on streams will now also preserve the stream identifier as they are queued on the opposite channel. Finally when a stream topology change is requested it is immediately accepted and reflected on both channels. Each channel also receives a queued frame to indicate that the topology has changed. ASTERISK-28938 Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186	6 years ago
Kevin Harwell	3d1bf3c537	Compiler fixes for gcc 10 ... This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9	6 years ago
Ben Ford	3927f79cb5	res_stir_shaken: Add inbound INVITE support. ... Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an INVITE, the Identity header is retrieved, parsing the message to verify the signature. If any of the parsing fails, AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this caller ID. If verification itself fails, AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in the payload does not line up with the SIP signaling, AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the verification process. A new config option has been added to the general section for stir_shaken.conf. "signature_timeout" is the amount of time a signature will be considered valid. If an INVITE is received and the amount of time between when it was received and when it was signed is greater than signature_timeout, verification will fail. Some changes were also made to signing and verification. There was an error where the whole JSON string was being signed rather than the header combined with the payload. This has been changed to sign the correct thing. Verification has been changed to do this as well, and the unit tests have been updated to reflect these changes. A couple of utility functions have also been added. One decodes a BASE64 string and returns the decoded string, doing all the length calculations for you. The other retrieves a string value from a header in a rdata object. Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913	6 years ago
George Joseph	ca3c22c5f1	Scope Tracing: A new facility for tracing scope enter/exit ... What's wrong with ast_debug? ast_debug is fine for general purpose debug output but it's not really geared for scope tracing since it doesn't present its output in a way that makes capturing and analyzing flow through Asterisk easy. How is scope tracing better? Scope tracing uses the same "cleanup" attribute that RAII_VAR uses to print messages to a separate "trace" log level. Even better, the messages are indented and unindented based on a thread-local call depth counter. When output to a separate log file, the output is uncluttered and easy to follow. Here's an example of the output. The leading timestamps and thread ids are removed and the output cut off at 68 columns for commit message restrictions but you get the idea. --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 --> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 --> res_pjsip_session.c:3669 handle_incoming_response PJSIP/ --> chan_pjsip.c:3265 chan_pjsip_incoming_response_after --> chan_pjsip.c:3194 chan_pjsip_incoming_response P chan_pjsip.c:3245 chan_pjsip_incoming_respon <-- chan_pjsip.c:3194 chan_pjsip_incoming_response P <-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after <-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/ <-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 The messages with the "-->" or "<--" were produced by including the following at the top of each function: SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session)); Scope isn't limited to functions any more than RAII_VAR is. You can also see entry and exit from "if", "for", "while", etc blocks. There is also an ast_trace() macro that doesn't track entry or exit but simply outputs a message to the trace log using the current indent level. The deepest message in the sample (chan_pjsip.c:3245) was used to indicate which "case" in a "select" was executed. How do you use it? More documentation is available in logger.h but here's an overview: * Configure with --enable-dev-mode. Like debug, scope tracing is #ifdef'd out if devmode isn't enabled. * Add a SCOPE_TRACE() call to the top of your function. * Set a logger channel in logger.conf to output the "trace" level. * Use the CLI (or cli.conf) to set a trace level similar to setting debug level... CLI> core set trace 2 res_pjsip.so Summary Of Changes: * Added LOG_TRACE logger level. Actually it occupies the slot formerly occupied by the now defunct "event" level. * Added core asterisk option "trace" similar to debug. Includes ability to specify global trace level in asterisk.conf and CLI commands to turn on/off and set levels. Levels can be set globally (probably not a good idea), or by module/source file. * Updated sample asterisk.conf and logger.conf. Tracing is disabled by default in both. * Added __ast_trace() to logger.c which keeps track of the indent level using TLS. It's #ifdef'd out if devmode isn't enabled. * Added ast_trace() and SCOPE_TRACE() macros to logger.h. These are all #ifdef'd out if devmode isn't enabled. Why not use gcc's -finstrument-functions capability? gcc's facility doesn't allow access to local data and doesn't operate on non-function scopes. Known Issues: The only know issue is that we currently don't know the line number where the scope exited. It's reported as the same place the scope was entered. There's probably a way to get around it but it might involve looking at the stack and doing an 'addr2line' to get the line number. Kind of like ast_backtrace() does. Not sure if it's worth it. Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027	6 years ago
Ben Ford	e29df34de0	res_stir_shaken: Added dialplan function and API call. ... Adds the "STIR_SHAKEN" dialplan function and an API call to add a STIR_SHAKEN verification result to a channel. This information will be held in a datastore on the channel that can later be queried through the "STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results including identity, attestation, and verify_result. Here are some examples: STIR_SHAKEN(count) STIR_SHAKEN(0, identity) STIR_SHAKEN(1, attestation) STIR_SHAKEN(2, verify_result) Getting the count can be used to iterate through the results and pull information by specifying the index and the field you want to retrieve. Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82	6 years ago
Nathan Bruning	f217fcdc62	app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions ... Add a new "masquarade" channel event, and use it in app_queue to track unique id's. Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210 ASTERISK-28829 #close ASTERISK-25844 #close Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6	6 years ago
Ben Ford	9acf840f7c	res_stir_shaken: Implemented signature verification. ... There are a lot of moving parts in this patch, but the focus of it is on the verification of the signature using a public key located at the public key URL provided in the JSON payload. First, we check the database to see if we have already downloaded the key. If so, check to see if it has expired. If it has, redownload from the URL. If we don't have an entry in the database, just go ahead and download the public key. The expiration is tested each time we download the file. After that, read the public key from the file and use it to verify the signature. All sanity checking is done when the payload is first received, so the verification is complete once this point is reached. The XML has also been added since a new config option was added to general (curl_timeout). The maximum amount of time to wait for a download can be configured through this option, with a low value by default. Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c	6 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
Joshua C. Colp	6cfc6ff53c	confbridge: Add support for disabling text messaging. ... When in a conference bridge it may be necessary to have text messages disabled for specific participants or for all. This change adds a configuration option, "text_messaging", which can be used to enable or disable this on the user profile. By default existing behavior is preserved as it defaults to "yes". ASTERISK-28841 Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13	6 years ago
Jean Aunis	de66713fd5	func_volume: Accept decimal number as argument ... Allow voice volume to be multiplied or divided by a floating point number. ASTERISK-28813 Change-Id: I5b42b890ec4e1f6b0b3400cb44ff16522b021c8c	6 years ago
George Joseph	7ba6d43083	test_res_pjsip_session_caps: Create unit test ... This unit test runs through combinations of... * Local codecs * Remote Codecs * Codec Preference * Incoming/Outgoing A few new APIs were created to make it easier to test the functionality but didn't result in any actual functional change. ASTERISK_28777 Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164	6 years ago
George Joseph	2ee455958e	codec_negotiation: Implement outgoing_call_offer_pref ... Based on this new endpoint setting, a joint list of preferred codecs between those received from the Asterisk core (remote), and those specified in the endpoint's "allow" parameter (local) is created and is used to create the outgoing SDP offer. * Add outgoing_call_offer_pref to pjsip_configuration (endpoint) * Add "call_direction" to res_pjsip_session. * Update pjsip_session_caps.c to make the functions more generic so they could be used for both incoming and outgoing. * Update ast_sip_session_create_outgoing to create the pending_media_state->topology with the results of ast_sip_session_create_joint_call_stream(). * The endpoint "preferred_codec_only" option now automatically sets AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref. * A helper function ast_stream_get_format_count() was added to streams to return the current count of formats. ASTERISK-28777 Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437	6 years ago
Ben Ford	57a457c26c	res_stir_shaken: Implemented signing of JSON payload. ... This change provides functions that take in a JSON payload, verify that the contents contain all the mandatory fields and required values (if any), and signs the payload with the private key. Four fields are added to the payload: x5u, attest, iat, and origid. As of now, these are just placeholder values that will be set to actual values once the logic is implemented for what to do when an actual payload is received, but the functions to add these values have all been implemented and are ready to use. Upon successful signing and the addition of those four values, a ast_stir_shaken_payload is returned, containing other useful information such as the algorithm and signature. Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393	6 years ago
Jaco Kroon	d32e559e8a	acl: implement a centralized ACL output mechanism for HAs and ACLs. ... named_acl.c (which is really a named_ha) now uses ast_ha_output. I've also updated main/manager.c to output the actual ACL on "manager show user <username>" if one is set. If this works then we can add similar to other modules as required. Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f	6 years ago
Jaco Kroon	6f731f153b	netsock2: compile fixes. ... This fixes ast_addressfamily_to_sockaddrsize to reference the provided argument, and ast_sockaddr_from_sockaddr to not use the name of a structure as argument. Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7	6 years ago
Ben Ford	211bb8a79c	res_stir_shaken: Initial commit and reading private key. ... This commit sets up some of the initial framework for the module and adds a way to read the private key from the specified file, which will then be appended to the certificate object. This works fine for now, but eventually some other structure will likely need to be used to store all this information. Similarly, the caller_id_number is specified on the certificate config object, but in the end we will want that information to be tied to the certificate itself and read it from there. A method has been added that will retrieve the private key associated with the caller_id_number passed in. Tab completion for certificates and stores has also been added. Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f	6 years ago
Sean Bright	d68f940f6e	dns_txt: Add TXT record parsing support ... Change-Id: Ie0eca23b8e6f4c7d9846b6013d79099314d90ef5	6 years ago
George Joseph	99efe1f868	Merge "codec negotiation: add incoming_call_offer_prefs option"	6 years ago
Joshua Colp	bdf7b4eeb3	Merge "message & stasis/messaging: make text message variables work in ARI"	6 years ago
Kevin Harwell	06dada3f01	codec negotiation: add incoming_call_offer_prefs option ... Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that specifies the preferred order of codecs after receiving an offer. This patch does the following: Adds a new enumeration, ast_sip_call_codec_pref, used by the the new configuration option that's added to the endpoint media structure. Adds a new ast_sip_session_caps structure that's set for each session media object. Creates a new file, res_pjsip_session_caps that "implements" the new structure and option, and is compiled into the res_pjsip_session library. ASTERISK-28756 #close Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f	6 years ago
Kevin Harwell	f8a852605d	Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions"	6 years ago
Kevin Harwell	a715cf5aaa	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	6 years ago
Kevin Harwell	d18af40431	Merge "say: Remove unused "plural" option from main/say"	6 years ago
Kevin Harwell	566f9a541f	Merge "format_cap: make function parameters 'const'"	6 years ago
Kevin Harwell	a3b3a9d2dc	Merge "pjsip: Update ACLs on named ACL changes."	6 years ago
Torrey Searle	77c9ba8e63	res/res_pjsip_sdp_rtp: Fix MOH transitions ... Update the state of remote_hold immediately on receipt of remote SDP so that the information is available when building the SDP answer ASTERISK-28754 #close Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f	6 years ago
Kevin Harwell	1e1651b4f4	format_cap: make function parameters 'const' ... There were a couple places where the format cap function parameter was not 'const' when it should have been. This patch makes them 'const'. Change-Id: Ife753fb16a962d842a6b44f45363a61a66bfdb2e	6 years ago
Walter Doekes	0b5c6fddf1	say: Remove unused "plural" option from main/say ... There are exceptions for plural objects, but they are detected using the supplied NUMBER, not using an extra option. Change-Id: I95d1d1b2796b1aba92048a2dbae8a3856ed8a113	6 years ago
George Joseph	3854b561a5	Merge "bridging: Add better support for adding/removing streams."	6 years ago
Joshua C. Colp	d6712790cd	pjsip: Update ACLs on named ACL changes. ... This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	6 years ago
Joshua C. Colp	5a5be92b79	bridging: Add better support for adding/removing streams. ... This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	6 years ago
Ben Ford	168637cc0c	RTP/ICE: Send on first valid pair. ... When handling ICE negotiations, it's possible that there can be a delay between STUN binding requests which in turn will cause a delay in ICE completion, preventing media from flowing. It should be possible to send media when there is at least one valid pair, preventing this scenario from occurring. A change was added to PJPROJECT that adds an optional callback (on_valid_pair) that will be called when the first valid pair is found during ICE negotiation. Asterisk uses this to start the DTLS handshake, allowing media to flow. It will only be called once, either on the first valid pair, or when ICE negotiation is complete. ASTERISK-28716 Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867	6 years ago
George Joseph	b76ab5e5c9	message.c: Add option to suppress the Message channel AMI and ARI events ... In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" preserving existing behavior. Beginning with Asterisk 18, the option will default to "yes". NOTE: This change does not affect UserEvents or the ARI TextMessageReceived events. * Added the "hide_messaging_ami_events" option to asterisk.conf. * Changed message.c to set the AST_CHAN_TP_INTERNAL property on the "Message/ast_msg_queue" channel if the option is set in asterisk.conf. This suppresses the reporting of the events. Change-Id: Ia2e3516d43f4e0df994fc6598565d6bba2d7018b	6 years ago
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	6 years ago
Friendly Automation	3f663a543d	Merge "netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr."	6 years ago
Seán C McCord	163efbd724	feat: AudioSocket channel, application, and ARI support. ... This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5	6 years ago
Friendly Automation	51f811183a	Merge "ARI: Ability to inhibit COLP frames when adding channels to a bridge"	6 years ago
Jaco Kroon	3bc8b36537	netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr. ... ast_addressfamily_to_sockaddrize will determine the size that's required, and ast_sockaddr_from_sockaddr then wraps this new function and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without knowing the address family) into the ast_sockaddr structure. Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Friendly Automation	34746220a0	Merge "res_pjsip_pubsub: Add ability to persist generator state information."	6 years ago
Joshua Colp	a55d403429	Merge "res_pjsip_endpoint_identifier_ip.c: Add port matching support"	6 years ago
Joshua C. Colp	4e7adbd8f4	res_pjsip_pubsub: Add ability to persist generator state information. ... Some body generators, such as dialog-info+xml, require storing state information which is then conveyed in the NOTIFY request itself. Up until now there was no way for such body generators to persist this information. Two new API calls have been added to allow body generators to set and get persisted data. This data is persisted out alongside the normal persistence information and allows the body generator to restore state information or to simply use this for normal storage of state. State is stored in the form of JSON and it is up to the body generator to interpret this as needed. The dialog-info+xml body generator has been updated to take advantage of this to persist the version number. ASTERISK-27759 Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de	6 years ago
Joshua Colp	bf0247ae7c	Merge "stasis.c: Use correct topic name in stasis_topic_pool_delete_topic"	6 years ago
Sean Bright	312abaa1fe	res_pjsip_endpoint_identifier_ip.c: Add port matching support ... Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	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
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
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	07b7c6f50a	Merge "confbridge: Add support for specifying maximum sample rate."	6 years ago