34131 Commits (371493087ae20b7a0ad3f9e05eebdbacbba6da0a)

 

All Branches   

Search

Author	SHA1	Message	Date
Sean Bright	371493087a	channelstorage_makeopts.xml: Remove errant XML character. ... Resolves: #1282	1 year ago
mkmer	e63d87a064	utils.h: Add rounding to float conversion to int. ... Quote from an audio engineer NR9V: There is a minor issue of a small amount of crossover distortion though as a result of `ast_slinear_saturated_multiply_float()` not rounding the float. This could result in some quiet but potentially audible distortion artifacts in lower volume parts of the signal. If you have for example a sign wave function with a max amplitude of just a few samples, all samples between -1 and 1 will be truncated to zero, resulting in the waveform no longer being a sine wave and in harmonic distortion. Resolves: #1176	1 year ago
Tinet-mucw	678f721e4d	pbx.c: when set flag AST_SOFTHANGUP_ASYNCGOTO, ast_explicit_goto should return -1. ... Under certain circumstances the context/extens/prio are set in the ast_async_goto, for example action Redirect. In the situation that action Redirect is broken by GotoIf this info is changed. that will causes confusion in dialplan execution. Resolves: #1273	1 year ago
Sean Bright	41409669d4	res_musiconhold.c: Ensure we're always locked around music state access.	1 year ago
Sean Bright	855152eb79	res_musiconhold.c: Annotate when the channel is locked.	1 year ago
Jaco Kroon	23d2d37b66	res_musiconhold: Appropriately lock channel during start. ... This relates to #829 This doesn't sully solve the Ops issue, but it solves the specific crash there. Further PRs to follow. In the specific crash the generator was still under construction when moh was being stopped, which then proceeded to close the stream whilst it was still in use. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	1 year ago
George Joseph	ec2591c60b	res_stir_shaken.so: Handle X5U certificate chains. ... The verification process will now load a full certificate chain retrieved via the X5U URL instead of loading only the end user cert. * Renamed crypto_load_cert_from_file() and crypto_load_cert_from_memory() to crypto_load_cert_chain_from_file() and crypto_load_cert_chain_from_memory() respectively. * The two load functions now continue to load certs from the file or memory PEMs and store them in a separate stack of untrusted certs specific to the current verification context. * crypto_is_cert_trusted() now uses the stack of untrusted certs that were extracted from the PEM in addition to any untrusted certs that were passed in from the configuration (and any CA certs passed in from the config of course). Resolves: #1272 UserNote: The STIR/SHAKEN verification process will now load a full certificate chain retrieved via the X5U URL instead of loading only the end user cert.	1 year ago
George Joseph	6e9c33caad	res_stir_shaken: Add "ignore_sip_date_header" config option. ... UserNote: A new STIR/SHAKEN verification option "ignore_sip_date_header" has been added that when set to true, will cause the verification process to not consider a missing or invalid SIP "Date" header to be a failure. This will make the IAT the sole "truth" for Date in the verification process. The option can be set in the "verification" and "profile" sections of stir_shaken.conf. Also fixed a bug in the port match logic. Resolves: #1251 Resolves: #1271	1 year ago
Naveen Albert	47250b716c	app_record: Add RECORDING_INFO function. ... Add a function that can be used to retrieve info about a previous recording, such as its duration. This is being added as a function to avoid possibly trampling on dialplan variables, and could be extended to provide other information in the future. Resolves: #548 UserNote: The RECORDING_INFO function can now be used to retrieve the duration of a recording.	1 year ago
Itzanh	30e7209249	app_sms.c: Fix sending and receiving SMS messages in protocol 2 ... This fixes bugs in SMS messaging to SMS-capable analog phones that prevented app_sms.c from talking to phones using SMS protocol 2. - Fix MORX message reception (from phone to Asterisk) in SMS protocol 2 - Fix MTTX message transmission (from Asterisk to phone) in SMS protocol 2 One of the bugs caused messages to have random characters and junk appended at the end up to the character limit. Another bug prevented Asterisk from sending messages from Asterisk to the phone at all. A final bug caused the transmission from Asterisk to the phone to take a long time because app_sms.c did not hang up after correctly sending the message, causing the phone to have to time out and hang up in order to complete the message transmission. This was tested with a Linksys PAP2T and with a GrandStream HT814, sending and receiving messages with Telefónica DOMO Mensajes phones from Telefónica Spain. I had to play with both the network jitter buffer and the dB gain to get it to work. One of my phones required the gain to be set to +3dB for it to work, while another required it to be set to +6dB. Only MORX and MTTX were tested, I did not test sending and receiving messages to a TelCo SMSC.	1 year ago
phoneben	12440d232f	app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penalties only for members within min/max range ... This update adds support for a new QUEUE_RAISE_PENALTY format: rN When QUEUE_RAISE_PENALTY is set to rN (e.g., r4), only members whose current penalty is greater than or equal to the defined min_penalty and less than or equal to max_penalty will have their penalty raised to N. Members with penalties outside the min/max range remain unchanged. Example behaviors: QUEUE_RAISE_PENALTY=4 → Raise all members with penalty < 4 (existing behavior) QUEUE_RAISE_PENALTY=r4 → Raise only members with penalty in [min_penalty, max_penalty] to 4 Implementation details: Adds parsing logic to detect the r prefix and sets the raise_respect_min flag Modifies the raise logic to skip members outside the defined penalty range when the flag is active UserNote: This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective penalty raises only for members whose current penalty is within the [min_penalty, max_penalty] range. Members with lower or higher penalties are unaffected. This behavior is backward-compatible with existing queue rule configurations.	1 year ago
George Joseph	a743003e1e	res_websocket_client: Add more info to the XML documentation. ... Added "see-also" links to chan_websocket and ARI Outbound WebSocket and added an example configuration for each.	1 year ago
Jaco Kroon	e125410e5c	res_odbc: cache_size option to limit the cached connections. ... Signed-off-by: Jaco Kroon <jaco@uls.co.za> UserNote: New cache_size option for res_odbc to on a per class basis limit the number of cached connections. Please reference the sample configuration for details.	1 year ago
Jaco Kroon	49d87e4f81	res_odbc: cache_type option for res_odbc. ... This enables setting cache_type classes to a round-robin queueing system rather than the historic stack mechanism. This should result in lower risk of connection drops due to shorter idle times (the first connection to go onto the stack could in theory never be used again, ever, but sit there consuming resources, there could be multiple of these). And with a queue rather than a stack, dead connections are guaranteed to be detected and purged eventually. This should end up better balancing connection_cnt with actual load over time, assuming the database doesn't keep connections open excessively long from it's side. Signed-off-by: Jaco Kroon <jaco@uls.co.za> UserNote: When using res_odbc it should be noted that back-end connections to the underlying database can now be configured to re-use the cached connections in a round-robin manner rather than repeatedly re-using the same connection. This helps to keep connections alive, and to purge dead connections from the system, thus more dynamically adjusting to actual load. The downside is that one could keep too many connections active for a longer time resulting in resource also begin consumed on the database side.	1 year ago
Sean Bright	b1065e783d	res_pjsip: Fix empty `ActiveChannels` property in AMI responses. ... The logic appears to have been reversed since it was introduced in 05cbf8df. Resolves: #1254	1 year ago
George Joseph	1c0d552155	ARI Outbound Websockets ... Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws Code change summary: * Added an ast_vector_string_join() function, * Added ApplicationRegistered and ApplicationUnregistered ARI events. * Converted res/ari/config.c to use sorcery to process ari.conf. * Added the "outbound-websocket" ARI config object. * Refactored res/ari/ari_websockets.c to handle outbound websockets. * Refactored res/ari/cli.c for the sorcery changeover. * Updated res/res_stasis.c for the sorcery changeover. * Updated apps/app_stasis.c to allow initiating per-call outbound websockets. * Added CLI commands to manage ARI websockets. * Added the new "outbound-websocket" object to ari.conf.sample. * Moved the ARI XML documentation out of res_ari.c into res/ari/ari_doc.xml UserNote: Asterisk can now establish websocket sessions _to_ your ARI applications as well as accepting websocket sessions _from_ them. Full details: http://s.asterisk.net/ari-outbound-ws	1 year ago
George Joseph	36fc358bc9	res_websocket_client: Create common utilities for websocket clients. ... Since multiple Asterisk capabilities now need to create websocket clients it makes sense to create a common set of utilities rather than making each of those capabilities implement their own. * A new configuration file "websocket_client.conf" is used to store common client parameters in named configuration sections. * APIs are provided to list and retrieve ast_websocket_client objects created from the named configurations. * An API is provided that accepts an ast_websocket_client object, connects to the remote server with retries and returns an ast_websocket object. TLS is supported as is basic authentication. * An observer can be registered to receive notification of loaded or reloaded client objects. * An API is provided to compare an existing client object to one just reloaded and return the fields that were changed. The caller can then decide what action to take based on which fields changed. Also as part of thie commit, several sorcery convenience macros were created to make registering common object fields easier. UserNote: A new module "res_websocket_client" and config file "websocket_client.conf" have been added to support several upcoming new capabilities that need common websocket client configuration.	1 year ago
George Joseph	fe1ab659ad	asterisk.c: Add option to restrict shell access from remote consoles. ... UserNote: A new asterisk.conf option 'disable_remote_console_shell' has been added that, when set, will prevent remote consoles from executing shell commands using the '!' prefix. Resolves: #GHSA-c7p6-7mvq-8jq2	1 year ago
George Joseph	1b8b17f150	res_pjsip_messaging.c: Mask control characters in received From display name ... Incoming SIP MESSAGEs will now have their From header's display name sanitized by replacing any characters < 32 (space) with a space. Resolves: #GHSA-2grh-7mhv-fcfw	1 year ago
mkmer	113c7d0a8d	frame.c: validate frame data length is less than samples when adjusting volume ... Resolves: #1230	1 year ago
Sven Kube	0f414f6a94	res_audiosocket.c: Add retry mechanism for reading data from AudioSocket ... The added retry mechanism addresses an issue that arises when fragmented TCP packets are received, each containing only a portion of an AudioSocket packet. This situation can occur if the external service sending the AudioSocket data has Nagle's algorithm enabled.	1 year ago
Sven Kube	c0e8f4f63b	res_audiosocket.c: Set the TCP_NODELAY socket option ... Disable Nagle's algorithm by setting the TCP_NODELAY socket option. This reduces latency by preventing delays caused by packet buffering.	1 year ago
Thomas B. Clark	28f5d4a2ec	menuselect: Fix GTK menu callbacks for Fedora 42 compatibility ... This patch resolves a build failure in `menuselect_gtk.c` when running `make menuconfig` on Fedora 42. The new version of GTK introduced stricter type checking for callback signatures. Changes include: - Add wrapper functions to match the expected `void (*)(void)` signature. - Update `menu_items` array to use these wrappers. Fixes: #1243	1 year ago
Stanislav Abramenkov	296cf69925	jansson: Upgrade version to jansson 2.14.1 ... UpgradeNote: jansson has been upgraded to 2.14.1. For more information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.14.1 Resolves: #1178	1 year ago
Joe Searle	ae5ea528ca	pjproject: Increase maximum SDP formats and attribute limits ... Since Chrome 136, using Windows, when initiating a video call the INVITE SDP exceeds the maximum number of allowed attributes, resulting in the INVITE being rejected. This increases the attribute limit and the number of formats allowed when using bundled pjproject. Fixes: #1240	1 year ago
Nathan Monfils	019d4ef17c	manager.c: Invalid ref-counting when purging events ... We have a use-case where we generate a *lot* of events on the AMI, and then when doing `manager show eventq` we would see some events which would linger for hours or days in there. Obviously something was leaking. Testing allowed us to track down this logic bug in the ref-counting on the event purge. Reproducing the bug was not super trivial, we managed to do it in a production-like load testing environment with multiple AMI consumers. The race condition itself: 1. something allocates and links `session` 2. `purge_sessions` iterates over that `session` (takes ref) 3. `purge_session` correctly de-referencess that session 4. `purge_session` re-evaluates the while() loop, taking a reference 5. `purge_session` exits (`n_max > 0` is false) 6. whatever allocated the `session` deallocates it, but a reference is now lost since we exited the `while` loop before de-referencing. 7. since the destructor is never called, the session->last_ev->usecount is never decremented, leading to events lingering in the queue The impact of this bug does not seem major. The events are small and do not seem, from our testing, to be causing meaningful additional CPU usage. Mainly we wanted to fix this issue because we are internally adding prometheus metrics to the eventq and those leaked events were causing the metrics to show garbage data.	1 year ago
Mike Bradeen	0a24944001	res_pjsip_nat.c: Do not overwrite transfer host ... When a call is transfered via dialplan behind a NAT, the host portion of the Contact header in the 302 will no longer be over-written with the external NAT IP and will retain the hostname. Fixes: #1141	1 year ago
Mike Bradeen	a5ac74ef68	chan_pjsip: Serialize INVITE creation on DTMF attended transfer ... When a call is transfered via DTMF feature code, the Transfer Target and Transferer are bridged immediately. This opens the possibilty of a race condition between the creation of an INVITE and the bridge induced colp update that can result in the set caller ID being over-written with the transferer's default info. Fixes: #1234	1 year ago
George Joseph	8146faad58	Alternate Channel Storage Backends ... Full details: http://s.asterisk.net/dc679ec3 The previous proof-of-concept showed that the cpp_map_name_id alternate storage backed performed better than all the others so this final PR adds only that option. You still need to enable it in menuselect under the "Alternate Channel Storage Backends" category. To select which one is used at runtime, set the "channel_storage_backend" option in asterisk.conf to one of the values described in asterisk.conf.sample. The default remains "ao2_legacy". UpgradeNote: With this release, you can now select an alternate channel storage backend based on C++ Maps. Using the new backend may increase performance and reduce the chances of deadlocks on heavily loaded systems. For more information, see http://s.asterisk.net/dc679ec3	1 year ago
Naveen Albert	876c25a953	sig_analog: Add Call Waiting Deluxe support. ... Adds support for Call Waiting Deluxe options to enhance the current call waiting feature. As part of this change, a mechanism is also added that allows a channel driver to queue an audio file for Dial() to play, which is necessary for the announcement function. ASTERISK-30373 #close Resolves: #271 UserNote: Call Waiting Deluxe can now be enabled for FXS channels by enabling its corresponding option.	1 year ago
Naveen Albert	dc2d559ccf	app_sms: Ignore false positive vectorization warning. ... Ignore gcc warning about writing 32 bytes into a region of size 6, since we check that we don't go out of bounds for each byte. This is due to a vectorization bug in gcc 15, stemming from gcc commit 68326d5d1a593dc0bf098c03aac25916168bc5a9. Resolves: #1088	1 year ago
George Joseph	54682a538a	lock.h: Add include for string.h when DEBUG_THREADS is defined. ... When DEBUG_THREADS is defined, lock.h uses strerror(), which is defined in the libc string.h file, to print warning messages. If the including source file doesn't include string.h then strerror() won't be found and and compile errors will be thrown. Since lock.h depends on this, string.h is now included from there if DEBUG_THREADS is defined. This way, including source files don't have to worry about it.	1 year ago
Naveen Albert	2bb607f7b7	res_pjsip_caller_id: Also parse URI parameters for ANI2. ... If the isup-oli was sent as a URI parameter, rather than a header parameter, it was not being parsed. Make sure we parse both if needed so the ANI2 is set regardless of which type of parameter the isup-oli is sent as. Resolves: #1220	1 year ago
Naveen Albert	be9c2cd6ff	app_meetme: Remove inaccurate removal version from xmldocs. ... app_meetme is deprecated but wasn't removed as planned in 21, so remove the inaccurate removal version. Resolves: #1224	1 year ago
Luz Paz	a20cfc68ef	docs: Fix typos in apps/ ... Found via codespell	1 year ago
Mike Bradeen	4dc3ca4c9a	stasis/control.c: Set Hangup Cause to No Answer on Dial timeout ... Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs, but the Dial command via ARI did not set an explicit reason. This resulted in a CANCEL with Normal Call Clearing and corresponding ChannelDestroyed. This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the other operations. Fixes: #963 UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.	1 year ago
Naveen Albert	06f8092ae9	chan_iax2: Minor improvements to documentation and warning messages. ... * Update Dial() documentation for IAX2 to include syntax for RSA public key names. * Add additional details to a couple warnings to provide more context when an undecodable frame is received. Resolves: #1206	1 year ago
Andreas Wehrmann	c00e809ff0	pbx_ael: unregister AELSub application and CLI commands on module load failure ... This fixes crashes/hangs I noticed with Asterisk 20.3.0 and 20.13.0 and quickly found out, that the AEL module doesn't do proper cleanup when it fails to load. This happens for example when there are syntax errors and AEL fails to compile in which case pbx_load_module() returns an error but load_module() doesn't then unregister CLI cmds and the application.	1 year ago
Albrecht Oster	c251afadb9	res_pjproject: Fix DTLS client check failing on some platforms ... Certain platforms (mainly BSD derivatives) have an additional length field in `sockaddr_in6` and `sockaddr_in`. `ast_sockaddr_from_pj_sockaddr()` does not take this field into account when copying over values from the `pj_sockaddr` into the `ast_sockaddr`. The resulting `ast_sockaddr` will have an uninitialized value for `sin6_len`/`sin_len` while the other `ast_sockaddr` (not converted from a `pj_sockaddr`) to check against in `ast_sockaddr_pj_sockaddr_cmp()` has the correct length value set. This has the effect that `ast_sockaddr_cmp()` will always indicate an address mismatch, because it does a bitwise comparison, and all DTLS packets are dropped even if addresses and ports match. `ast_sockaddr_from_pj_sockaddr()` now checks whether the length fields are available on the current platform and sets the values accordingly. Resolves: #505	1 year ago
George Joseph	f8bc3ddeb9	Prequisites for ARI Outbound Websockets ... stasis: * Added stasis_app_is_registered(). * Added stasis_app_control_mark_failed(). * Added stasis_app_control_is_failed(). * Fixed res_stasis_device_state so unsubscribe all works properly. * Modified stasis_app_unregister() to unsubscribe from all event sources. * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() returns true. http: * Added ast_http_create_basic_auth_header(). md5: * Added define for MD5_DIGEST_LENGTH. tcptls: * Added flag to ast_tcptls_session_args to suppress connection log messages to give callers more control over logging. http_websocket: * Add flag to ast_websocket_client_options to suppress connection log messages to give callers more control over logging. * Added username and password to ast_websocket_client_options to support outbound basic authentication. * Added ast_websocket_result_to_str().	1 year ago
Ben Ford	bff3fd0fa8	contrib: Add systemd service and timer files for malloc trim. ... Adds two files to the contrib/systemd/ directory that can be installed to periodically run "malloc trim" on Asterisk. These files do nothing unless they are explicitly moved to the correct location on the system. Users who are experiencing Asterisk memory issues can use this service to potentially help combat the problem. These files can also be configured to change the start time and interval. See systemd.timer(5) and systemd.time(7) for more information. UserNote: Service and timer files for systemd have been added to the contrib/systemd/ directory. If you are experiencing memory issues, install these files to have "malloc trim" periodically run on the system.	1 year ago
Peter Jannesen	6881b6249f	action_redirect: remove after_bridge_goto_info ... Under certain circumstances the context/extens/prio are stored in the after_bridge_goto_info. This info is used when the bridge is broken by for hangup of the other party. In the situation that the bridge is broken by an AMI Redirect this info is not used but also not removed. With the result that when the channel is put back in a bridge and the bridge is broken the execution continues at the wrong context/extens/prio. Resolves: #1144	1 year ago
Joshua C. Colp	bcd0e53ef6	channel: Always provide cause code in ChannelHangupRequest. ... When queueing a channel to be hung up a cause code can be specified in one of two ways: 1. ast_queue_hangup_with_cause This function takes in a cause code and queues it as part of the hangup request, which ultimately results in it being set on the channel. 2. ast_channel_hangupcause_set + ast_queue_hangup This combination sets the hangup cause on the channel before queueing the hangup instead of as part of that process. In the #2 case the ChannelHangupRequest event would not contain the cause code. For consistency if a cause code has been set on the channel it will now be added to the event. Resolves: #1197	1 year ago
phoneben	7457d7d215	Add log-caller-id-name option to log Caller ID Name in queue log ... Add log-caller-id-name option to log Caller ID Name in queue log This patch introduces a new global configuration option, log-caller-id-name, to queues.conf to control whether the Caller ID name is logged when a call enters a queue. When log-caller-id-name=yes, the Caller ID name is logged as parameter 4 in the queue log, provided it’s allowed by the existing log_restricted_caller_id rules. If log-caller-id-name=no (the default), the Caller ID name is omitted from the logs. Fixes: #1091 UserNote: This patch adds a global configuration option, log-caller-id-name, to queues.conf to control whether the Caller ID name is logged as parameter 4 when a call enters a queue. When log-caller-id-name=yes, the Caller ID name is included in the queue log, Any '|' characters in the caller ID name will be replaced with '_'. (provided it’s allowed by the existing log_restricted_caller_id rules). When log-caller-id-name=no (the default), the Caller ID name is omitted.	1 year ago
George Joseph	ade69af6d9	asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf. ... Commands in the "[startup_commands]" section of cli.conf have historically run after all core and module initialization has been completed and just before "Asterisk Ready" is printed on the console. This meant that if you wanted to debug initialization of a specific module, your only option was to turn on debug for everything by setting "debug" in asterisk.conf. This commit introduces options to allow you to run CLI commands earlier in the asterisk startup process. A command with a value of "pre-init" will run just after logger initialization but before most core, and all module, initialization. A command with a value of "pre-module" will run just after all core initialization but before all module initialization. A command with a value of "fully-booted" (or "yes" for backwards compatibility) will run as they always have been...after all initialization and just before "Asterisk Ready" is printed on the console. This means you could do this... ``` [startup_commands] core set debug 3 res_pjsip.so = pre-module core set debug 0 res_pjsip.so = fully-booted ``` This would turn debugging on for res_pjsip.so to catch any module initialization debug messages then turn it off again after the module is loaded. UserNote: In cli.conf, you can now define startup commands that run before core initialization and before module initialization.	1 year ago
Sean Bright	8bae6a1d8c	app_confbridge: Prevent crash when publishing channel-less event. ... Resolves: #1190	1 year ago
George Joseph	62e73f9bd8	ari_websockets: Fix frack if ARI config fails to load. ... ari_ws_session_registry_dtor() wasn't checking that the container was valid before running ao2_callback on it to shutdown registered sessions.	2 years ago
George Joseph	6bc055416b	ARI: REST over Websocket ... This commit adds the ability to make ARI REST requests over the same websocket used to receive events. For full details on how to use the new capability, visit... https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ Changes: * Added utilities to http.c: * ast_get_http_method_from_string(). * ast_http_parse_post_form(). * Added utilities to json.c: * ast_json_nvp_array_to_ast_variables(). * ast_variables_to_json_nvp_array(). * Added definitions for new events to carry REST responses. * Created res/ari/ari_websocket_requests.c to house the new request handlers. * Moved non-event specific code out of res/ari/resource_events.c into res/ari/ari_websockets.c * Refactored res/res_ari.c to move non-http code out of ast_ari_callback() (which is http specific) and into ast_ari_invoke() so it can be shared between both the http and websocket transports. UpgradeNote: This commit adds the ability to make ARI REST requests over the same websocket used to receive events. See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/	2 years ago
mkmer	ca8adc2454	audiohook.c: Add ability to adjust volume with float ... Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41. This is accomplished by the following: Convert internal variables to type float. Always use ast_frame_adjust_volume_float() for adjustments. Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). Cast float to int in ast_audiohook_volume_get() Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float. This update maintains 100% backward compatibility. Resolves: #1171	2 years ago
Florent CHAUVEAU	ea657ec7c7	audiosocket: added support for DTMF frames ... Updated the AudioSocket protocol to allow sending DTMF frames. AST_FRAME_DTMF frames are now forwarded to the server, in addition to AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF with value 0x03 was added to the protocol. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...). UserNote: The AudioSocket protocol now forwards DTMF frames with payload type 0x03. The payload is a 1-byte ascii representing the DTMF digit (0-9,*,#...).	2 years ago