From 1814a4620d88cce42d1b5aeb5e3e68014f03493f Mon Sep 17 00:00:00 2001 From: Tristan Matthews Date: Fri, 6 Jan 2012 19:24:42 -0500 Subject: [PATCH] callmanager/confmanager: cleanup Fixed whitespace, removed unused methods, corrected documentation. --- daemon/src/dbus/callmanager-introspec.xml | 1470 ++++++++--------- daemon/src/dbus/callmanager.cpp | 11 +- daemon/src/dbus/callmanager.h | 1 - .../dbus/configurationmanager-introspec.xml | 1202 +++++++------- gnome/src/dbus/callmanager-introspec.xml | 1470 ++++++++--------- .../dbus/configurationmanager-introspec.xml | 1202 +++++++------- kde/src/dbus/callmanager-introspec.xml | 1470 ++++++++--------- .../dbus/configurationmanager-introspec.xml | 1202 +++++++------- 8 files changed, 3943 insertions(+), 4085 deletions(-) diff --git a/daemon/src/dbus/callmanager-introspec.xml b/daemon/src/dbus/callmanager-introspec.xml index 46cec4006..5c38193ff 100644 --- a/daemon/src/dbus/callmanager-introspec.xml +++ b/daemon/src/dbus/callmanager-introspec.xml @@ -1,825 +1,781 @@ - + - -

The CallManager interface is used to manage - any call and conference related actions.

-

Since SFLphone-daemon support multiple incoming/outgoing calls, any actions involving a specific call must address the method by the means of a unique callID. SFLphone-clients is responsible to generate the callID on outgoing call. On the other hand, SFLphone-daemon will generate a unique callID on incoming calls.

- - - -

This is the main method in order to place a new call. The call is registered to the daemon using this method.

- - - - The ID of the account you want to make a call with. If the call is to be placed whithout any account by the means of a SIP URI (i.e. sip:num@server), the "IP2IP_PROFILE" is passed as the accountID. For more details about accounts see the configuration manager interface. - - - - - The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. - - - - - If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. - - - + +

The CallManager interface is used to manage call and conference related actions.

+

Since SFLphone-daemon supports multiple incoming/outgoing calls, any actions involving a specific call must address the method by the means of a unique callID. + SFLphone-clients is responsible for generating the callID on outgoing calls. Conversely, SFLphone-daemon will generate a unique callID for incoming calls.

+ + + +

This is the main method in order to place a new call. The call is registered with the daemon using this method.

+ + + + The ID of the account with which you want to make a call. If the call is to be placed without any account by means of a SIP URI (i.e. sip:num@server), the "IP2IP_PROFILE" is passed as the accountID. For more details on accounts see the configuration manager interface. + + + + + The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. + + + + + If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. + + + - - - - Place a call with the fist registered account, regarding to the account list order. - - Use this function when you don't have any information about the accounts used (Ex: Firefly mozilla extension) - - - - - The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. - - - - - If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. - - - + + + + Place a call with the first registered account in the account list. + + Use this function when you don't have any information about the accounts used (Ex: Firefly mozilla extension) + + + + + The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. + + + + + If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. + + + - - - Refuse an incoming call. - - - - The callID. - - + + + Refuse an incoming call. + + + + The callID. + + - + - - - Answer an incoming call. Automatically put the current call on state HOLD. - - - - The callID. - - - + + + Answer an incoming call. Automatically puts the current call on HOLD. + + + + The callID. + + + - - - Hangup a call in state "CURRENT" or "HOLD". - - - - The callID. - - - + + + Hangup a call in state "CURRENT" or "HOLD". + + + + The callID. + + + - - - - Hangup a conference, and every call participating to the conference. - - - - The unique conference ID. - - - + + + + Hangup a conference, and every call participating to the conference. + + + + The unique conference ID. + + + - - - Place a call on hold. - - - - The callID. - - - + + + Place a call on hold. + + + + The callID. + + + - - - Hold off a call, and place this call on state CURRENT. - - - - The callID. - - - + + + Take a call off hold, and place this call in state CURRENT. + + + + The callID. + + + - - - Transfer a call to given phone number. - - - - The callID. - - - - - The phone number to transfer the call to. - - - - - - - Perform an attended transfer on two calls - - - - The callID of the call to be transfered. - - - - - The callID of the target call. - - - + + + Transfer a call to the given phone number. + + + + The callID. + + + + + The phone number to which the call will be transferred. + + + - - - Dual-Tone multi-frequency. Tell the core to play dial tones. A SIP INFO message is sent to notify the server. - - - - Unicode charter for pressed key - - - + + + Perform an attended transfer on two calls. + + + + The callID of the call to be transfered. + + + + + The callID of the target call. + + + - - - Start audio stream and play tone.. - - - - + + + Dual-Tone multi-frequency. Tell the core to play dialtones. A SIP INFO message is sent to notify the server. + + + + Unicode character for pressed key. + + + - - -

Sets the volume using a linear scale [0,100].

- Pulseaudio has its own mechanism to modify application volume. This method is enabled only if the ALSA API is used. - - - - The device: mic or speaker - - - - - The volume value (between 0 and 100) - - - + + + Start audio stream and play tone. + + + + - - -

Return the volume value of the given device on a linear scale [0,100].

- Only enabled if the ALSA API is used, Pulseaudio has its own mechanism to modify application volume. - - - - The device: mic or speaker - - - - - The volume value (between 0 and 100) - - - + + +

Sets the volume using a linear scale [0,100].

+ Pulseaudio has its own mechanism to modify application volume. This method is enabled only if the ALSA API is used. + + + + The device: mic or speaker + + + + + The volume value (between 0 and 100) + + + - - - -

Join two participants together to create a 3-way conference including the current client.

- The signal conferenceCreated is emitted on success. - - - - + + +

Return the volume value of the given device on a linear scale [0,100].

+ Only enabled if the ALSA API is used, Pulseaudio has its own mechanism to modify application volume. + + + + The device: mic or speaker + + + + + The volume value (between 0 and 100) + + + + + + + +

Join two participants together to create a 3-way conference including the current client.

+ The signal conferenceCreated is emitted on success. + + + + - - -

Create a conference from a list of participant

- The signal conferenceCreated is emitted on success. - - - + + +

Create a conference from a list of participants

+ The signal conferenceCreated is emitted on success. + + + - - - -

Join a new particiant to an existing conference.

- The signal conferenceChanged is emitted on success. - - - - The ID of the call to add to the conference - - - - - An existing conference ID - - - + + + +

Join a new particiant to an existing conference.

+ The signal conferenceChanged is emitted on success. + + + + The ID of the call to add to the conference + + + + + An existing conference ID + + + - - - -

As the core can handle multiple calls an conferences, it may happens that the client's user leave a conference to answer an incoming call or send new ones. This method is used to reintroduce SFLphone-client's user into the conference.

-

It put the current call on state HOLD or detach SFLphone-client's user from the another conference.

- - - - An existing conference ID - - - + + + +

As the core can handle multiple calls and conferences, it may happen that the client's user leaves a conference to answer an incoming call or to start new calls. This method is used to reintroduce SFLphone-client's user into the conference.

+

Its put the current call on HOLD or detaches SFLphone-client's user from the another conference.

+ + + + An existing conference ID + + + - - - - Detach the given call from the conference. If only one participant is left, the conference is deleted and the signal conferenceRemoved is emited. - - - - The call ID - - - + + + + Detach the given call from the conference. If only one participant is left, the conference is deleted and the signal conferenceRemoved is emited. + + + + The call ID + + + - - - - Join two conferences together. - - - - + + + + Join two conferences together. + + + + - - - - Returns a hashtable containing conference details. - - - - The conference ID - - - - - - A map containing the ID of the conferences - and their states: -
    -
  • ACTIVE_ATTACHED
    • -
  • ACTIVE_DETACHED
    • -
  • HOLD
    • -
- - - + + + + Returns a hashtable containing conference details. + + + + The conference ID + + + + + + A map containing the ID of the conferences + and their states: +
    +
  • ACTIVE_ATTACHED
    • +
  • ACTIVE_DETACHED
    • +
  • HOLD
    • +
+ + + - - - - Returns a list containing all active - conferences. - To update client status, one should - use getParticipantList - with provided conference IDs. - - - - The list of conferences. - - - + + + + Returns a list containing all active + conferences. + To update client status, one should + use getParticipantList + with provided conference IDs. + + + + The list of conferences. + + + - - - Start recording a call. - - - - The ID of the call to record. - - - + + + Start recording a call. + + + + The ID of the call to record. + + + - - - Tells whether or not a call is being recorded. - - - - The call ID. - - - - - Returns true is the call is being recorded. False otherwise. - - - + + + Tells whether or not a call is being recorded. + + + + The call ID. + + + + + Returns true is the call is being recorded. False otherwise. + + + - - - Once after starting recording for the first time, this signal is emited to - provide the recorded file path to client application. - - - - + + + Once after starting recording for the first time, this signal is emited to + provide the recorded file path to client application. + + + + - - - - - + + + + - - - Get all the details about a specific call. - - - - The call ID. - - - - - -

A map containing the call details:

-
    -
  • ACCOUNTID
    • -
  • PEER_NUMBER
    • -
  • PEER_NAME
    • -
  • DISPLAY_NAME
    • -
  • CALL_STATE
    • -
  • CALL_TYPE
    • -
- - - - - - Get the list of active calls. - To get the call details, iterate on the return value and call getCallDetails method. - - - - A list of call IDs. - - - + + + Get all the details about a specific call. + + + + The call ID. + + + + + +

A map containing the call details:

+
    +
  • ACCOUNTID
    • +
  • PEER_NUMBER
    • +
  • PEER_NAME
    • +
  • DISPLAY_NAME
    • +
  • CALL_STATE
    • +
  • CALL_TYPE
    • +
+ + + - - - Unused - - - - - - + + + Get the list of active calls. + To get the call details, iterate on the return value and call getCallDetails method. + + + + A list of call IDs. + + + - - - Unused - - - - + + + + - - - Send a text message to the specified call - - - - + + + Send a text message to the specified call + + + + - - -

Notify that a cell have been created.

-

The callID generated by the daemon must be stored by the clients in order to address other action for - this call. This signal is emitted when call have been created by the daemon itself.

- The client must subscribe to this signal to handle calls created by other clients - - - - The account ID of the calle. Clients must notify teh right account when receiving this signal. - - - - - A new call ID. - - - - - The sip uri this call is trying to reach - - - + + +

Notify that a call has been created.

+

The callID generated by the daemon must be stored by the clients in order to address other actions for + this call. This signal is emitted when call haves been created by the daemon itself.

+ The client must subscribe to this signal to handle calls created by other clients + + + + The account ID of the call. Clients must notify the right account when receiving this signal. + + + + + A new call ID. + + + + + The SIP URI this call is trying to reach. + + + - - -

Notify an incoming call.

-

The callID generated by the daemon must be stored by the clients in order to address other action for - this call. This signal is emitted when we receive a call from a remote peer

- The client must subscribe to this signal to handle incoming calls. - - - - The account ID of the callee. Clients must notify the right account when receiving this signal. - - - - - A new call ID. - - - - - The caller phone number. - - - + + +

Notify an incoming call.

+

The callID generated by the daemon must be stored by the clients in order to address other action for + this call. This signal is emitted when we receive a call from a remote peer

+ The client must subscribe to this signal to handle incoming calls. + + + + The account ID of the callee. Clients must notify the right account when receiving this signal. + + + + + A new call ID. + + + + + The caller phone number. + + + - - - Notify clients that a new text message has been received. - - - - - + + + Notify clients that a new text message has been received. + + + + + - - -

Notify of a change in a call state.

-

The client must subscribe to this signal.

- - - - The call ID. - - - - - The acceptable states are: -
    -
  • INCOMING: Initial state of incoming calls
    • -
  • RINGING: Initial state of received outgoing call
    • -
  • CURRENT: The normal active state of an answered call
    • -
  • HUNGUP: Notify that the call has been hungup by peer
    • -
  • BUSY
    • -
  • FAILURE: Error when processing a call
    • -
  • HOLD
    • -
  • UNHOLD_CURRENT
    • -
  • UNHOLD_RECORD
    • -
- - - + + +

Notify of a change in a call state.

+

The client must subscribe to this signal.

+ + + + The call ID. + + + + + The acceptable states are: +
    +
  • INCOMING: Initial state of incoming calls
    • +
  • RINGING: Initial state of received outgoing call
    • +
  • CURRENT: The normal active state of an answered call
    • +
  • HUNGUP: Notify that the call has been hungup by peer
    • +
  • BUSY
    • +
  • FAILURE: Error when processing a call
    • +
  • HOLD
    • +
  • UNHOLD_CURRENT
    • +
  • UNHOLD_RECORD
    • +
+ + + - - - - Notify of a change in the conferences state - - - - The conference ID. - - - - - The acceptable states are: -
    -
  • ACTIVE_ATTACHED: SFLphone user is - participating to this conference
    • -
  • ACTIVE_DETACHED: This situation can - occur if a call is received while - SFLphone user is participating to a - conference. In this case, one can leave - the conference by answering the - call. Other participants may continue - conferencing normally.
    • -
  • HOLD: Each call in this conference - is on state HOLD
    • -
- - - + + + + Notify of a change in the conferences state + + + + The conference ID. + + + + + The acceptable states are: +
    +
  • ACTIVE_ATTACHED: SFLphone user is + participating to this conference
    • +
  • ACTIVE_DETACHED: This situation can + occur if a call is received while + SFLphone user is participating to a + conference. In this case, one can leave + the conference by answering the + call. Other participants may continue + conferencing normally.
    • +
  • HOLD: Each call in this conference + is on state HOLD
    • +
+ + + - - - - Get the call IDs of every participant to a given conference. The client should keep and update the list of participant. - - - - The conference ID. - - - - - The list of the call IDs. - - - + + + + Get the call IDs of every participant to a given conference. The client should keep and update the list of participants. + + + + The conference ID. + + + + + The list of the call IDs. + + + - - - - Emited when a new conference is created. SFLphone-client is reponsible to store the confID and call getParticipantList to update the display. - - - - A new conference ID. - - - + + + + Emited when a new conference is created. SFLphone-client is reponsible for storing the confID and call getParticipantList to update the display. + + + + A new conference ID. + + + - - - - Emited when a new conference is remove. SFLphone-client should have kept a list of current participant in order to display modification. - - - - The conference ID. - - - + + + + Emited when a new conference is remove. SFLphone-client should have kept a list of current participant in order to display modification. + + + + The conference ID. + + + - - - - Hold on every calls participating to this conference. - - - - The conference ID. - - - + + + + Hold every call which is participating in this conference. + + + + The conference ID. + + + - - - - Hold off every calls participating to this conference. - - - - The conference ID. - - - - - - - - - - - + + + + Hold off every call participating in this conference. + + + + The conference ID. + + + - - - - - + + + + + - - -

Call state changed, SFLphone received a notification - from registrar concerning this call.

- - - - The call ID - - - - Description string - - - - The SIP or IAX2 message code - - + + + + + - - -

Account state changed, SFLphone received a notification - from registrar.

- - - - The account ID - - - - - Description string - - - - - The SIP or IAX2 message code - - - + + +

Call state changed, SFLphone received a notification + from registrar concerning this call.

+ + + + The call ID + + + + Description string + + + + The SIP or IAX2 message code + + - - - Notify the clients of the voicemail number for a specific account, if applicable. - - - - The account ID. - - - - - The number of waiting messages. - - - + + +

Account state changed, SFLphone received a notification + from registrar.

+ + + + The account ID + + + + + Description string + + + + + The SIP or IAX2 message code + + + - - -

Notify clients of a volume level - change.

-

This signal occurs only if ALSA is - enabled since Pulseaudio streams are - managed externally.

- - - - The device: mic or speaker - - - - - The new volume value - - - + + + Notify the clients of the voicemail number for a specific account, if applicable. + + + + The account ID. + + + + + The number of waiting messages. + + + - - -

Transfer has been successfully - processed. Client should remove transfered - call from call list as it is no longer - accessible in SFLphone-daemon.

- - + + +

Notify clients of a volume level change.

+

This signal occurs only if ALSA is enabled since Pulseaudio streams are managed externally.

+ + + + The device: mic or speaker + + + + + The new volume value + + + - - -

Transfer operation failed. Corespondin - call is no longer accessible in - SFLphone-daemon.

- - + + +

Transfer has been successfully + processed. Client should remove transfered + call from call list as it is no longer + accessible in SFLphone-daemon.

+ + - - - -

Signal sent on SDES session success. Media transmission is encripted - for this call only. It does not apply for a - conference.

-

A conference can be considered to be secured if and only if each - participant is secured.

- - - + + +

Transfer operation failed. Corespondin + call is no longer accessible in + SFLphone-daemon.

+ + - - - -

Sinal sent to notify that SDES session - failed.

-

Media transmission is not encrypted.

- - - + + + +

Signal sent on SDES session success. Media transmission is encripted + for this call only. It does not apply for a conference.

+

A conference can be considered to be secured if and only if each + participant is secured.

+ + + - - - - - - - - + + + +

Sinal sent to notify that SDES session failed.

+

Media transmission is not encrypted.

+ + + - - - - - - + + + + + + - - - - - - + + + + - - - - - - - - + + + + - - - - - - + + + + + + - - - - - - - - - + + + + - - - - - - + + + + + + + - - - - - - + + + + - - - - - - + + + + - - - - - - + + + + - - - - - - - + + + + - - - - - - - + + + + + - + + + + + + + diff --git a/daemon/src/dbus/callmanager.cpp b/daemon/src/dbus/callmanager.cpp index 3a13295ef..809e0628e 100644 --- a/daemon/src/dbus/callmanager.cpp +++ b/daemon/src/dbus/callmanager.cpp @@ -237,14 +237,11 @@ CallManager::getIsRecording(const std::string& callID) return Manager::instance().isRecording(callID); } - -std::string -CallManager::getCurrentAudioCodecName(const std::string& callID) +std::string CallManager::getCurrentAudioCodecName(const std::string& callID) { return Manager::instance().getCurrentCodecName(callID).c_str(); } - std::map CallManager::getCallDetails(const std::string& callID) { @@ -257,12 +254,6 @@ CallManager::getCallList() return Manager::instance().getCallList(); } -std::string -CallManager::getCurrentCallID() -{ - return Manager::instance().getCurrentCallId(); -} - void CallManager::playDTMF(const std::string& key) { diff --git a/daemon/src/dbus/callmanager.h b/daemon/src/dbus/callmanager.h index e0592f32b..7758fa723 100644 --- a/daemon/src/dbus/callmanager.h +++ b/daemon/src/dbus/callmanager.h @@ -86,7 +86,6 @@ class CallManager void attendedTransfer(const std::string& transferID, const std::string& targetID); std::map< std::string, std::string > getCallDetails(const std::string& callID); std::vector< std::string > getCallList(); - std::string getCurrentCallID(); /* Conference related methods */ void joinParticipant(const std::string& sel_callID, const std::string& drag_callID); diff --git a/daemon/src/dbus/configurationmanager-introspec.xml b/daemon/src/dbus/configurationmanager-introspec.xml index d86ffc2b6..5f30f2526 100644 --- a/daemon/src/dbus/configurationmanager-introspec.xml +++ b/daemon/src/dbus/configurationmanager-introspec.xml @@ -1,107 +1,107 @@ - + - - Used to handle the configuration stuff: accounts settings, account registration, user preferences, ... - + + Used to handle the configuration stuff: accounts settings, account registration, user preferences, ... + - - - Get all parameters of the specified account. - - - - The account ID - - - - - - The available keys / parameters are: -
    -
  • CONFIG_ACCOUNT_ENABLE: True or False (Default: True)
    • -
  • CONFIG_ACCOUNT_RESOLVE_ONCE
    • -
  • CONFIG_ACCOUNT_TYPE: SIP or IAX2 (Default: SIP)
    • -
  • HOSTNAME: The IP adress or hostname of the registrar
    • -
  • USERNAME: The username (or extension) of the account
    • -
  • PASSWORD: The password associated to the account
    • -
  • REALM
    • -
  • CONFIG_ACCOUNT_MAILBOX: Number to dial to access the voicemail box
    • -
  • CONFIG_ACCOUNT_REGISTRATION_EXPIRE: SIP header expiration value (Default: 1600)
    • -
  • LOCAL_INTERFACE: The network interface (Default: eth0)
    • -
  • PUBLISHED_SAMEAS_LOCAL: If False, the published address equals the local address. This is the default.
    • -
  • PUBLISHED_ADDRESS: The SIP published address
    • -
  • LOCAL_PORT: The SIP listening port (Default: 5060)
    • -
  • PUBLISHED_PORT: The SIP published port
    • -
  • DISPLAY_NAMEL: The display name
    • -
  • STUN_ENABLE: True or False (Default: False)
    • -
  • STUN_SERVER: The STUN server address
    • -
  • REGISTRATION_STATUS: The account registration status. Should be Registered to make calls.
    • -
  • REGISTRATION_STATE_CODE
    • -
  • REGISTRATION_STATE_DESCRIPTION
    • -
  • SRTP_KEY_EXCHANGE
    • -
  • SRTP_ENABLE: Whether or not voice communication are encrypted - True or False (Default: False)
    • -
  • SRTP_RTP_FALLBACK
    • -
  • ZRTP_DISPLAY_SAS
    • -
  • ZRTP_DISPLAY_SAS_ONCE
    • -
  • ZRTP_HELLO_HASH
    • -
  • ZRTP_NOT_SUPP_WARNING
    • -
  • TLS_LISTENER_PORT: TLS listening port (Default: 5061)
    • -
  • TLS_ENABLE: Whether or not signalling is encrypted - True or False (Default: False)
    • -
  • TLS_CA_LIST_FILE
    • -
  • TLS_CERTIFICATE_FILE
    • -
  • TLS_PRIVATE_KEY_FILE
    • -
  • TLS_METHOD
    • -
  • TLS_CIPHERS
    • -
  • TLS_SERVER_NAME
    • -
  • TLS_VERIFY_SERVER
    • -
  • TLS_VERIFY_CLIENT
    • -
  • TLS_REQUIRE_CLIENT_CERTIFICATE
    • -
  • TLS_NEGOTIATION_TIMEOUT_SEC
    • -
  • TLS_NEGOTIATION_TIMEOUT_MSEC
    • -
- - - + + + Get all parameters of the specified account. + + + + The account ID + + + + + + The available keys / parameters are: +
    +
  • CONFIG_ACCOUNT_ENABLE: True or False (Default: True)
    • +
  • CONFIG_ACCOUNT_RESOLVE_ONCE
    • +
  • CONFIG_ACCOUNT_TYPE: SIP or IAX2 (Default: SIP)
    • +
  • HOSTNAME: The IP adress or hostname of the registrar
    • +
  • USERNAME: The username (or extension) of the account
    • +
  • PASSWORD: The password associated to the account
    • +
  • REALM
    • +
  • CONFIG_ACCOUNT_MAILBOX: Number to dial to access the voicemail box
    • +
  • CONFIG_ACCOUNT_REGISTRATION_EXPIRE: SIP header expiration value (Default: 1600)
    • +
  • LOCAL_INTERFACE: The network interface (Default: eth0)
    • +
  • PUBLISHED_SAMEAS_LOCAL: If False, the published address equals the local address. This is the default.
    • +
  • PUBLISHED_ADDRESS: The SIP published address
    • +
  • LOCAL_PORT: The SIP listening port (Default: 5060)
    • +
  • PUBLISHED_PORT: The SIP published port
    • +
  • DISPLAY_NAMEL: The display name
    • +
  • STUN_ENABLE: True or False (Default: False)
    • +
  • STUN_SERVER: The STUN server address
    • +
  • REGISTRATION_STATUS: The account registration status. Should be Registered to make calls.
    • +
  • REGISTRATION_STATE_CODE
    • +
  • REGISTRATION_STATE_DESCRIPTION
    • +
  • SRTP_KEY_EXCHANGE
    • +
  • SRTP_ENABLE: Whether or not voice communication are encrypted - True or False (Default: False)
    • +
  • SRTP_RTP_FALLBACK
    • +
  • ZRTP_DISPLAY_SAS
    • +
  • ZRTP_DISPLAY_SAS_ONCE
    • +
  • ZRTP_HELLO_HASH
    • +
  • ZRTP_NOT_SUPP_WARNING
    • +
  • TLS_LISTENER_PORT: TLS listening port (Default: 5061)
    • +
  • TLS_ENABLE: Whether or not signalling is encrypted - True or False (Default: False)
    • +
  • TLS_CA_LIST_FILE
    • +
  • TLS_CERTIFICATE_FILE
    • +
  • TLS_PRIVATE_KEY_FILE
    • +
  • TLS_METHOD
    • +
  • TLS_CIPHERS
    • +
  • TLS_SERVER_NAME
    • +
  • TLS_VERIFY_SERVER
    • +
  • TLS_VERIFY_CLIENT
    • +
  • TLS_REQUIRE_CLIENT_CERTIFICATE
    • +
  • TLS_NEGOTIATION_TIMEOUT_SEC
    • +
  • TLS_NEGOTIATION_TIMEOUT_MSEC
    • +
+ + + - - - Send new account parameters, or account parameters changes, to the core. The hash table is not required to be complete, only the updated parameters may be specified. - Account settings are written to the configuration file when sflphone properly quits. - After calling this method, the core will emit the signal accountsChanged with the updated data. The client must subscribe to this signal and use it to update its internal data structure. - - - - - - - - - - - + + + Send new account parameters, or account parameters changes, to the core. The hash table is not required to be complete, only the updated parameters may be specified. + Account settings are written to the configuration file when sflphone properly quits. + After calling this method, the core will emit the signal accountsChanged with the updated data. The client must subscribe to this signal and use it to update its internal data structure. + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - + Get configuration settings of the IP2IP_PROFILE. They are sligthly different from account settings since no VoIP accounts are involved. - + Available parameters are: @@ -134,584 +134,584 @@ - - - - - - - - - - - - - + + + + + + + + + + + + + - - - Add a new account. When created, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. - If no details are specified, the default parameters are used. - The core tries to register the account as soon it is created. - - - - - The new account settings - - - - - A new account ID - - - + + + Add a new account. When created, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. + If no details are specified, the default parameters are used. + The core tries to register the account as soon it is created. + + + + + The new account settings + + + + + A new account ID + + + - - - Update the accounts order. - When placing a call, the first registered account in the list is used. - - - - An ordered list of account IDs, delimited by '/' - - - + + + Update the accounts order. + When placing a call, the first registered account in the list is used. + + + + An ordered list of account IDs, delimited by '/' + + + - - - Remove an existing account. When removed, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. - - - - The account to remove, identified by its ID - - - + + + Remove an existing account. When removed, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. + + + + The account to remove, identified by its ID + + + - - - Get a list of all created accounts, as stored by the core. - - - - - A list of account IDs - - - + + + Get a list of all created accounts, as stored by the core. + + + + + A list of account IDs + + + - - - Send account registration (REGISTER) to the registrar. - - the account if expire=1, unregister if expire=0. + + + Send account registration (REGISTER) to the registrar. + + the account if expire=1, unregister if expire=0. - @param[in] input accountID - --> - - - The account ID - - - - -

To register, expire must be 1.

-

To un-register, expire must be 0.

- - - + @param[in] input accountID + --> + + + The account ID + + + + +

To register, expire must be 1.

+

To un-register, expire must be 0.

+ + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - + - + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - + - - - - - - - - - + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - + + + + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - + + + + + + + - - - - + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - + + + + + + - - - - - - + + + + + + - - - - - - - - + + + + + + + + - - - - + + + + - + - - - - - - - - + + + + + + + + - - - - + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - + - + - + - - + + - - - - - - + + + + + + - - - - - - - - - - + + + + + + + + + + - + - - - - - - - + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - + + + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - + - - - - - - - + + + + + + + - + - - - - - - - + + + + + + + - + diff --git a/gnome/src/dbus/callmanager-introspec.xml b/gnome/src/dbus/callmanager-introspec.xml index 46cec4006..5c38193ff 100644 --- a/gnome/src/dbus/callmanager-introspec.xml +++ b/gnome/src/dbus/callmanager-introspec.xml @@ -1,825 +1,781 @@ - + - -

The CallManager interface is used to manage - any call and conference related actions.

-

Since SFLphone-daemon support multiple incoming/outgoing calls, any actions involving a specific call must address the method by the means of a unique callID. SFLphone-clients is responsible to generate the callID on outgoing call. On the other hand, SFLphone-daemon will generate a unique callID on incoming calls.

- - - -

This is the main method in order to place a new call. The call is registered to the daemon using this method.

- - - - The ID of the account you want to make a call with. If the call is to be placed whithout any account by the means of a SIP URI (i.e. sip:num@server), the "IP2IP_PROFILE" is passed as the accountID. For more details about accounts see the configuration manager interface. - - - - - The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. - - - - - If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. - - - + +

The CallManager interface is used to manage call and conference related actions.

+

Since SFLphone-daemon supports multiple incoming/outgoing calls, any actions involving a specific call must address the method by the means of a unique callID. + SFLphone-clients is responsible for generating the callID on outgoing calls. Conversely, SFLphone-daemon will generate a unique callID for incoming calls.

+ + + +

This is the main method in order to place a new call. The call is registered with the daemon using this method.

+ + + + The ID of the account with which you want to make a call. If the call is to be placed without any account by means of a SIP URI (i.e. sip:num@server), the "IP2IP_PROFILE" is passed as the accountID. For more details on accounts see the configuration manager interface. + + + + + The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. + + + + + If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. + + + - - - - Place a call with the fist registered account, regarding to the account list order. - - Use this function when you don't have any information about the accounts used (Ex: Firefly mozilla extension) - - - - - The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. - - - - - If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. - - - + + + + Place a call with the first registered account in the account list. + + Use this function when you don't have any information about the accounts used (Ex: Firefly mozilla extension) + + + + + The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. + + + + + If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. + + + - - - Refuse an incoming call. - - - - The callID. - - + + + Refuse an incoming call. + + + + The callID. + + - + - - - Answer an incoming call. Automatically put the current call on state HOLD. - - - - The callID. - - - + + + Answer an incoming call. Automatically puts the current call on HOLD. + + + + The callID. + + + - - - Hangup a call in state "CURRENT" or "HOLD". - - - - The callID. - - - + + + Hangup a call in state "CURRENT" or "HOLD". + + + + The callID. + + + - - - - Hangup a conference, and every call participating to the conference. - - - - The unique conference ID. - - - + + + + Hangup a conference, and every call participating to the conference. + + + + The unique conference ID. + + + - - - Place a call on hold. - - - - The callID. - - - + + + Place a call on hold. + + + + The callID. + + + - - - Hold off a call, and place this call on state CURRENT. - - - - The callID. - - - + + + Take a call off hold, and place this call in state CURRENT. + + + + The callID. + + + - - - Transfer a call to given phone number. - - - - The callID. - - - - - The phone number to transfer the call to. - - - - - - - Perform an attended transfer on two calls - - - - The callID of the call to be transfered. - - - - - The callID of the target call. - - - + + + Transfer a call to the given phone number. + + + + The callID. + + + + + The phone number to which the call will be transferred. + + + - - - Dual-Tone multi-frequency. Tell the core to play dial tones. A SIP INFO message is sent to notify the server. - - - - Unicode charter for pressed key - - - + + + Perform an attended transfer on two calls. + + + + The callID of the call to be transfered. + + + + + The callID of the target call. + + + - - - Start audio stream and play tone.. - - - - + + + Dual-Tone multi-frequency. Tell the core to play dialtones. A SIP INFO message is sent to notify the server. + + + + Unicode character for pressed key. + + + - - -

Sets the volume using a linear scale [0,100].

- Pulseaudio has its own mechanism to modify application volume. This method is enabled only if the ALSA API is used. - - - - The device: mic or speaker - - - - - The volume value (between 0 and 100) - - - + + + Start audio stream and play tone. + + + + - - -

Return the volume value of the given device on a linear scale [0,100].

- Only enabled if the ALSA API is used, Pulseaudio has its own mechanism to modify application volume. - - - - The device: mic or speaker - - - - - The volume value (between 0 and 100) - - - + + +

Sets the volume using a linear scale [0,100].

+ Pulseaudio has its own mechanism to modify application volume. This method is enabled only if the ALSA API is used. + + + + The device: mic or speaker + + + + + The volume value (between 0 and 100) + + + - - - -

Join two participants together to create a 3-way conference including the current client.

- The signal conferenceCreated is emitted on success. - - - - + + +

Return the volume value of the given device on a linear scale [0,100].

+ Only enabled if the ALSA API is used, Pulseaudio has its own mechanism to modify application volume. + + + + The device: mic or speaker + + + + + The volume value (between 0 and 100) + + + + + + + +

Join two participants together to create a 3-way conference including the current client.

+ The signal conferenceCreated is emitted on success. + + + + - - -

Create a conference from a list of participant

- The signal conferenceCreated is emitted on success. - - - + + +

Create a conference from a list of participants

+ The signal conferenceCreated is emitted on success. + + + - - - -

Join a new particiant to an existing conference.

- The signal conferenceChanged is emitted on success. - - - - The ID of the call to add to the conference - - - - - An existing conference ID - - - + + + +

Join a new particiant to an existing conference.

+ The signal conferenceChanged is emitted on success. + + + + The ID of the call to add to the conference + + + + + An existing conference ID + + + - - - -

As the core can handle multiple calls an conferences, it may happens that the client's user leave a conference to answer an incoming call or send new ones. This method is used to reintroduce SFLphone-client's user into the conference.

-

It put the current call on state HOLD or detach SFLphone-client's user from the another conference.

- - - - An existing conference ID - - - + + + +

As the core can handle multiple calls and conferences, it may happen that the client's user leaves a conference to answer an incoming call or to start new calls. This method is used to reintroduce SFLphone-client's user into the conference.

+

Its put the current call on HOLD or detaches SFLphone-client's user from the another conference.

+ + + + An existing conference ID + + + - - - - Detach the given call from the conference. If only one participant is left, the conference is deleted and the signal conferenceRemoved is emited. - - - - The call ID - - - + + + + Detach the given call from the conference. If only one participant is left, the conference is deleted and the signal conferenceRemoved is emited. + + + + The call ID + + + - - - - Join two conferences together. - - - - + + + + Join two conferences together. + + + + - - - - Returns a hashtable containing conference details. - - - - The conference ID - - - - - - A map containing the ID of the conferences - and their states: -
    -
  • ACTIVE_ATTACHED
    • -
  • ACTIVE_DETACHED
    • -
  • HOLD
    • -
- - - + + + + Returns a hashtable containing conference details. + + + + The conference ID + + + + + + A map containing the ID of the conferences + and their states: +
    +
  • ACTIVE_ATTACHED
    • +
  • ACTIVE_DETACHED
    • +
  • HOLD
    • +
+ + + - - - - Returns a list containing all active - conferences. - To update client status, one should - use getParticipantList - with provided conference IDs. - - - - The list of conferences. - - - + + + + Returns a list containing all active + conferences. + To update client status, one should + use getParticipantList + with provided conference IDs. + + + + The list of conferences. + + + - - - Start recording a call. - - - - The ID of the call to record. - - - + + + Start recording a call. + + + + The ID of the call to record. + + + - - - Tells whether or not a call is being recorded. - - - - The call ID. - - - - - Returns true is the call is being recorded. False otherwise. - - - + + + Tells whether or not a call is being recorded. + + + + The call ID. + + + + + Returns true is the call is being recorded. False otherwise. + + + - - - Once after starting recording for the first time, this signal is emited to - provide the recorded file path to client application. - - - - + + + Once after starting recording for the first time, this signal is emited to + provide the recorded file path to client application. + + + + - - - - - + + + + - - - Get all the details about a specific call. - - - - The call ID. - - - - - -

A map containing the call details:

-
    -
  • ACCOUNTID
    • -
  • PEER_NUMBER
    • -
  • PEER_NAME
    • -
  • DISPLAY_NAME
    • -
  • CALL_STATE
    • -
  • CALL_TYPE
    • -
- - - - - - Get the list of active calls. - To get the call details, iterate on the return value and call getCallDetails method. - - - - A list of call IDs. - - - + + + Get all the details about a specific call. + + + + The call ID. + + + + + +

A map containing the call details:

+
    +
  • ACCOUNTID
    • +
  • PEER_NUMBER
    • +
  • PEER_NAME
    • +
  • DISPLAY_NAME
    • +
  • CALL_STATE
    • +
  • CALL_TYPE
    • +
+ + + - - - Unused - - - - - - + + + Get the list of active calls. + To get the call details, iterate on the return value and call getCallDetails method. + + + + A list of call IDs. + + + - - - Unused - - - - + + + + - - - Send a text message to the specified call - - - - + + + Send a text message to the specified call + + + + - - -

Notify that a cell have been created.

-

The callID generated by the daemon must be stored by the clients in order to address other action for - this call. This signal is emitted when call have been created by the daemon itself.

- The client must subscribe to this signal to handle calls created by other clients - - - - The account ID of the calle. Clients must notify teh right account when receiving this signal. - - - - - A new call ID. - - - - - The sip uri this call is trying to reach - - - + + +

Notify that a call has been created.

+

The callID generated by the daemon must be stored by the clients in order to address other actions for + this call. This signal is emitted when call haves been created by the daemon itself.

+ The client must subscribe to this signal to handle calls created by other clients + + + + The account ID of the call. Clients must notify the right account when receiving this signal. + + + + + A new call ID. + + + + + The SIP URI this call is trying to reach. + + + - - -

Notify an incoming call.

-

The callID generated by the daemon must be stored by the clients in order to address other action for - this call. This signal is emitted when we receive a call from a remote peer

- The client must subscribe to this signal to handle incoming calls. - - - - The account ID of the callee. Clients must notify the right account when receiving this signal. - - - - - A new call ID. - - - - - The caller phone number. - - - + + +

Notify an incoming call.

+

The callID generated by the daemon must be stored by the clients in order to address other action for + this call. This signal is emitted when we receive a call from a remote peer

+ The client must subscribe to this signal to handle incoming calls. + + + + The account ID of the callee. Clients must notify the right account when receiving this signal. + + + + + A new call ID. + + + + + The caller phone number. + + + - - - Notify clients that a new text message has been received. - - - - - + + + Notify clients that a new text message has been received. + + + + + - - -

Notify of a change in a call state.

-

The client must subscribe to this signal.

- - - - The call ID. - - - - - The acceptable states are: -
    -
  • INCOMING: Initial state of incoming calls
    • -
  • RINGING: Initial state of received outgoing call
    • -
  • CURRENT: The normal active state of an answered call
    • -
  • HUNGUP: Notify that the call has been hungup by peer
    • -
  • BUSY
    • -
  • FAILURE: Error when processing a call
    • -
  • HOLD
    • -
  • UNHOLD_CURRENT
    • -
  • UNHOLD_RECORD
    • -
- - - + + +

Notify of a change in a call state.

+

The client must subscribe to this signal.

+ + + + The call ID. + + + + + The acceptable states are: +
    +
  • INCOMING: Initial state of incoming calls
    • +
  • RINGING: Initial state of received outgoing call
    • +
  • CURRENT: The normal active state of an answered call
    • +
  • HUNGUP: Notify that the call has been hungup by peer
    • +
  • BUSY
    • +
  • FAILURE: Error when processing a call
    • +
  • HOLD
    • +
  • UNHOLD_CURRENT
    • +
  • UNHOLD_RECORD
    • +
+ + + - - - - Notify of a change in the conferences state - - - - The conference ID. - - - - - The acceptable states are: -
    -
  • ACTIVE_ATTACHED: SFLphone user is - participating to this conference
    • -
  • ACTIVE_DETACHED: This situation can - occur if a call is received while - SFLphone user is participating to a - conference. In this case, one can leave - the conference by answering the - call. Other participants may continue - conferencing normally.
    • -
  • HOLD: Each call in this conference - is on state HOLD
    • -
- - - + + + + Notify of a change in the conferences state + + + + The conference ID. + + + + + The acceptable states are: +
    +
  • ACTIVE_ATTACHED: SFLphone user is + participating to this conference
    • +
  • ACTIVE_DETACHED: This situation can + occur if a call is received while + SFLphone user is participating to a + conference. In this case, one can leave + the conference by answering the + call. Other participants may continue + conferencing normally.
    • +
  • HOLD: Each call in this conference + is on state HOLD
    • +
+ + + - - - - Get the call IDs of every participant to a given conference. The client should keep and update the list of participant. - - - - The conference ID. - - - - - The list of the call IDs. - - - + + + + Get the call IDs of every participant to a given conference. The client should keep and update the list of participants. + + + + The conference ID. + + + + + The list of the call IDs. + + + - - - - Emited when a new conference is created. SFLphone-client is reponsible to store the confID and call getParticipantList to update the display. - - - - A new conference ID. - - - + + + + Emited when a new conference is created. SFLphone-client is reponsible for storing the confID and call getParticipantList to update the display. + + + + A new conference ID. + + + - - - - Emited when a new conference is remove. SFLphone-client should have kept a list of current participant in order to display modification. - - - - The conference ID. - - - + + + + Emited when a new conference is remove. SFLphone-client should have kept a list of current participant in order to display modification. + + + + The conference ID. + + + - - - - Hold on every calls participating to this conference. - - - - The conference ID. - - - + + + + Hold every call which is participating in this conference. + + + + The conference ID. + + + - - - - Hold off every calls participating to this conference. - - - - The conference ID. - - - - - - - - - - - + + + + Hold off every call participating in this conference. + + + + The conference ID. + + + - - - - - + + + + + - - -

Call state changed, SFLphone received a notification - from registrar concerning this call.

- - - - The call ID - - - - Description string - - - - The SIP or IAX2 message code - - + + + + + - - -

Account state changed, SFLphone received a notification - from registrar.

- - - - The account ID - - - - - Description string - - - - - The SIP or IAX2 message code - - - + + +

Call state changed, SFLphone received a notification + from registrar concerning this call.

+ + + + The call ID + + + + Description string + + + + The SIP or IAX2 message code + + - - - Notify the clients of the voicemail number for a specific account, if applicable. - - - - The account ID. - - - - - The number of waiting messages. - - - + + +

Account state changed, SFLphone received a notification + from registrar.

+ + + + The account ID + + + + + Description string + + + + + The SIP or IAX2 message code + + + - - -

Notify clients of a volume level - change.

-

This signal occurs only if ALSA is - enabled since Pulseaudio streams are - managed externally.

- - - - The device: mic or speaker - - - - - The new volume value - - - + + + Notify the clients of the voicemail number for a specific account, if applicable. + + + + The account ID. + + + + + The number of waiting messages. + + + - - -

Transfer has been successfully - processed. Client should remove transfered - call from call list as it is no longer - accessible in SFLphone-daemon.

- - + + +

Notify clients of a volume level change.

+

This signal occurs only if ALSA is enabled since Pulseaudio streams are managed externally.

+ + + + The device: mic or speaker + + + + + The new volume value + + + - - -

Transfer operation failed. Corespondin - call is no longer accessible in - SFLphone-daemon.

- - + + +

Transfer has been successfully + processed. Client should remove transfered + call from call list as it is no longer + accessible in SFLphone-daemon.

+ + - - - -

Signal sent on SDES session success. Media transmission is encripted - for this call only. It does not apply for a - conference.

-

A conference can be considered to be secured if and only if each - participant is secured.

- - - + + +

Transfer operation failed. Corespondin + call is no longer accessible in + SFLphone-daemon.

+ + - - - -

Sinal sent to notify that SDES session - failed.

-

Media transmission is not encrypted.

- - - + + + +

Signal sent on SDES session success. Media transmission is encripted + for this call only. It does not apply for a conference.

+

A conference can be considered to be secured if and only if each + participant is secured.

+ + + - - - - - - - - + + + +

Sinal sent to notify that SDES session failed.

+

Media transmission is not encrypted.

+ + + - - - - - - + + + + + + - - - - - - + + + + - - - - - - - - + + + + - - - - - - + + + + + + - - - - - - - - - + + + + - - - - - - + + + + + + + - - - - - - + + + + - - - - - - + + + + - - - - - - + + + + - - - - - - - + + + + - - - - - - - + + + + + - + + + + + + + diff --git a/gnome/src/dbus/configurationmanager-introspec.xml b/gnome/src/dbus/configurationmanager-introspec.xml index d86ffc2b6..5f30f2526 100644 --- a/gnome/src/dbus/configurationmanager-introspec.xml +++ b/gnome/src/dbus/configurationmanager-introspec.xml @@ -1,107 +1,107 @@ - + - - Used to handle the configuration stuff: accounts settings, account registration, user preferences, ... - + + Used to handle the configuration stuff: accounts settings, account registration, user preferences, ... + - - - Get all parameters of the specified account. - - - - The account ID - - - - - - The available keys / parameters are: -
    -
  • CONFIG_ACCOUNT_ENABLE: True or False (Default: True)
    • -
  • CONFIG_ACCOUNT_RESOLVE_ONCE
    • -
  • CONFIG_ACCOUNT_TYPE: SIP or IAX2 (Default: SIP)
    • -
  • HOSTNAME: The IP adress or hostname of the registrar
    • -
  • USERNAME: The username (or extension) of the account
    • -
  • PASSWORD: The password associated to the account
    • -
  • REALM
    • -
  • CONFIG_ACCOUNT_MAILBOX: Number to dial to access the voicemail box
    • -
  • CONFIG_ACCOUNT_REGISTRATION_EXPIRE: SIP header expiration value (Default: 1600)
    • -
  • LOCAL_INTERFACE: The network interface (Default: eth0)
    • -
  • PUBLISHED_SAMEAS_LOCAL: If False, the published address equals the local address. This is the default.
    • -
  • PUBLISHED_ADDRESS: The SIP published address
    • -
  • LOCAL_PORT: The SIP listening port (Default: 5060)
    • -
  • PUBLISHED_PORT: The SIP published port
    • -
  • DISPLAY_NAMEL: The display name
    • -
  • STUN_ENABLE: True or False (Default: False)
    • -
  • STUN_SERVER: The STUN server address
    • -
  • REGISTRATION_STATUS: The account registration status. Should be Registered to make calls.
    • -
  • REGISTRATION_STATE_CODE
    • -
  • REGISTRATION_STATE_DESCRIPTION
    • -
  • SRTP_KEY_EXCHANGE
    • -
  • SRTP_ENABLE: Whether or not voice communication are encrypted - True or False (Default: False)
    • -
  • SRTP_RTP_FALLBACK
    • -
  • ZRTP_DISPLAY_SAS
    • -
  • ZRTP_DISPLAY_SAS_ONCE
    • -
  • ZRTP_HELLO_HASH
    • -
  • ZRTP_NOT_SUPP_WARNING
    • -
  • TLS_LISTENER_PORT: TLS listening port (Default: 5061)
    • -
  • TLS_ENABLE: Whether or not signalling is encrypted - True or False (Default: False)
    • -
  • TLS_CA_LIST_FILE
    • -
  • TLS_CERTIFICATE_FILE
    • -
  • TLS_PRIVATE_KEY_FILE
    • -
  • TLS_METHOD
    • -
  • TLS_CIPHERS
    • -
  • TLS_SERVER_NAME
    • -
  • TLS_VERIFY_SERVER
    • -
  • TLS_VERIFY_CLIENT
    • -
  • TLS_REQUIRE_CLIENT_CERTIFICATE
    • -
  • TLS_NEGOTIATION_TIMEOUT_SEC
    • -
  • TLS_NEGOTIATION_TIMEOUT_MSEC
    • -
- - - + + + Get all parameters of the specified account. + + + + The account ID + + + + + + The available keys / parameters are: +
    +
  • CONFIG_ACCOUNT_ENABLE: True or False (Default: True)
    • +
  • CONFIG_ACCOUNT_RESOLVE_ONCE
    • +
  • CONFIG_ACCOUNT_TYPE: SIP or IAX2 (Default: SIP)
    • +
  • HOSTNAME: The IP adress or hostname of the registrar
    • +
  • USERNAME: The username (or extension) of the account
    • +
  • PASSWORD: The password associated to the account
    • +
  • REALM
    • +
  • CONFIG_ACCOUNT_MAILBOX: Number to dial to access the voicemail box
    • +
  • CONFIG_ACCOUNT_REGISTRATION_EXPIRE: SIP header expiration value (Default: 1600)
    • +
  • LOCAL_INTERFACE: The network interface (Default: eth0)
    • +
  • PUBLISHED_SAMEAS_LOCAL: If False, the published address equals the local address. This is the default.
    • +
  • PUBLISHED_ADDRESS: The SIP published address
    • +
  • LOCAL_PORT: The SIP listening port (Default: 5060)
    • +
  • PUBLISHED_PORT: The SIP published port
    • +
  • DISPLAY_NAMEL: The display name
    • +
  • STUN_ENABLE: True or False (Default: False)
    • +
  • STUN_SERVER: The STUN server address
    • +
  • REGISTRATION_STATUS: The account registration status. Should be Registered to make calls.
    • +
  • REGISTRATION_STATE_CODE
    • +
  • REGISTRATION_STATE_DESCRIPTION
    • +
  • SRTP_KEY_EXCHANGE
    • +
  • SRTP_ENABLE: Whether or not voice communication are encrypted - True or False (Default: False)
    • +
  • SRTP_RTP_FALLBACK
    • +
  • ZRTP_DISPLAY_SAS
    • +
  • ZRTP_DISPLAY_SAS_ONCE
    • +
  • ZRTP_HELLO_HASH
    • +
  • ZRTP_NOT_SUPP_WARNING
    • +
  • TLS_LISTENER_PORT: TLS listening port (Default: 5061)
    • +
  • TLS_ENABLE: Whether or not signalling is encrypted - True or False (Default: False)
    • +
  • TLS_CA_LIST_FILE
    • +
  • TLS_CERTIFICATE_FILE
    • +
  • TLS_PRIVATE_KEY_FILE
    • +
  • TLS_METHOD
    • +
  • TLS_CIPHERS
    • +
  • TLS_SERVER_NAME
    • +
  • TLS_VERIFY_SERVER
    • +
  • TLS_VERIFY_CLIENT
    • +
  • TLS_REQUIRE_CLIENT_CERTIFICATE
    • +
  • TLS_NEGOTIATION_TIMEOUT_SEC
    • +
  • TLS_NEGOTIATION_TIMEOUT_MSEC
    • +
+ + + - - - Send new account parameters, or account parameters changes, to the core. The hash table is not required to be complete, only the updated parameters may be specified. - Account settings are written to the configuration file when sflphone properly quits. - After calling this method, the core will emit the signal accountsChanged with the updated data. The client must subscribe to this signal and use it to update its internal data structure. - - - - - - - - - - - + + + Send new account parameters, or account parameters changes, to the core. The hash table is not required to be complete, only the updated parameters may be specified. + Account settings are written to the configuration file when sflphone properly quits. + After calling this method, the core will emit the signal accountsChanged with the updated data. The client must subscribe to this signal and use it to update its internal data structure. + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - + Get configuration settings of the IP2IP_PROFILE. They are sligthly different from account settings since no VoIP accounts are involved. - + Available parameters are: @@ -134,584 +134,584 @@ - - - - - - - - - - - - - + + + + + + + + + + + + + - - - Add a new account. When created, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. - If no details are specified, the default parameters are used. - The core tries to register the account as soon it is created. - - - - - The new account settings - - - - - A new account ID - - - + + + Add a new account. When created, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. + If no details are specified, the default parameters are used. + The core tries to register the account as soon it is created. + + + + + The new account settings + + + + + A new account ID + + + - - - Update the accounts order. - When placing a call, the first registered account in the list is used. - - - - An ordered list of account IDs, delimited by '/' - - - + + + Update the accounts order. + When placing a call, the first registered account in the list is used. + + + + An ordered list of account IDs, delimited by '/' + + + - - - Remove an existing account. When removed, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. - - - - The account to remove, identified by its ID - - - + + + Remove an existing account. When removed, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. + + + + The account to remove, identified by its ID + + + - - - Get a list of all created accounts, as stored by the core. - - - - - A list of account IDs - - - + + + Get a list of all created accounts, as stored by the core. + + + + + A list of account IDs + + + - - - Send account registration (REGISTER) to the registrar. - - the account if expire=1, unregister if expire=0. + + + Send account registration (REGISTER) to the registrar. + + the account if expire=1, unregister if expire=0. - @param[in] input accountID - --> - - - The account ID - - - - -

To register, expire must be 1.

-

To un-register, expire must be 0.

- - - + @param[in] input accountID + --> + + + The account ID + + + + +

To register, expire must be 1.

+

To un-register, expire must be 0.

+ + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - + - + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - + - - - - - - - - - + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - + + + + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - + + + + + + + - - - - + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - + + + + + + - - - - - - + + + + + + - - - - - - - - + + + + + + + + - - - - + + + + - + - - - - - - - - + + + + + + + + - - - - + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - + - + - + - - + + - - - - - - + + + + + + - - - - - - - - - - + + + + + + + + + + - + - - - - - - - + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - + + + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - + - - - - - - - + + + + + + + - + - - - - - - - + + + + + + + - + diff --git a/kde/src/dbus/callmanager-introspec.xml b/kde/src/dbus/callmanager-introspec.xml index 46cec4006..5c38193ff 100755 --- a/kde/src/dbus/callmanager-introspec.xml +++ b/kde/src/dbus/callmanager-introspec.xml @@ -1,825 +1,781 @@ - + - -

The CallManager interface is used to manage - any call and conference related actions.

-

Since SFLphone-daemon support multiple incoming/outgoing calls, any actions involving a specific call must address the method by the means of a unique callID. SFLphone-clients is responsible to generate the callID on outgoing call. On the other hand, SFLphone-daemon will generate a unique callID on incoming calls.

- - - -

This is the main method in order to place a new call. The call is registered to the daemon using this method.

- - - - The ID of the account you want to make a call with. If the call is to be placed whithout any account by the means of a SIP URI (i.e. sip:num@server), the "IP2IP_PROFILE" is passed as the accountID. For more details about accounts see the configuration manager interface. - - - - - The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. - - - - - If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. - - - + +

The CallManager interface is used to manage call and conference related actions.

+

Since SFLphone-daemon supports multiple incoming/outgoing calls, any actions involving a specific call must address the method by the means of a unique callID. + SFLphone-clients is responsible for generating the callID on outgoing calls. Conversely, SFLphone-daemon will generate a unique callID for incoming calls.

+ + + +

This is the main method in order to place a new call. The call is registered with the daemon using this method.

+ + + + The ID of the account with which you want to make a call. If the call is to be placed without any account by means of a SIP URI (i.e. sip:num@server), the "IP2IP_PROFILE" is passed as the accountID. For more details on accounts see the configuration manager interface. + + + + + The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. + + + + + If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. + + + - - - - Place a call with the fist registered account, regarding to the account list order. - - Use this function when you don't have any information about the accounts used (Ex: Firefly mozilla extension) - - - - - The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. - - - - - If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. - - - + + + + Place a call with the first registered account in the account list. + + Use this function when you don't have any information about the accounts used (Ex: Firefly mozilla extension) + + + + + The callID is a unique identifier that must be randomly generated on the client's side. Any subsequent actions refering to this call must use this callID. + + + + + If bound to a VoIP account, then the argument is the phone number. In case of calls involving "IP2IP_PROFILE", a complete SIP URI must be specified. + + + - - - Refuse an incoming call. - - - - The callID. - - + + + Refuse an incoming call. + + + + The callID. + + - + - - - Answer an incoming call. Automatically put the current call on state HOLD. - - - - The callID. - - - + + + Answer an incoming call. Automatically puts the current call on HOLD. + + + + The callID. + + + - - - Hangup a call in state "CURRENT" or "HOLD". - - - - The callID. - - - + + + Hangup a call in state "CURRENT" or "HOLD". + + + + The callID. + + + - - - - Hangup a conference, and every call participating to the conference. - - - - The unique conference ID. - - - + + + + Hangup a conference, and every call participating to the conference. + + + + The unique conference ID. + + + - - - Place a call on hold. - - - - The callID. - - - + + + Place a call on hold. + + + + The callID. + + + - - - Hold off a call, and place this call on state CURRENT. - - - - The callID. - - - + + + Take a call off hold, and place this call in state CURRENT. + + + + The callID. + + + - - - Transfer a call to given phone number. - - - - The callID. - - - - - The phone number to transfer the call to. - - - - - - - Perform an attended transfer on two calls - - - - The callID of the call to be transfered. - - - - - The callID of the target call. - - - + + + Transfer a call to the given phone number. + + + + The callID. + + + + + The phone number to which the call will be transferred. + + + - - - Dual-Tone multi-frequency. Tell the core to play dial tones. A SIP INFO message is sent to notify the server. - - - - Unicode charter for pressed key - - - + + + Perform an attended transfer on two calls. + + + + The callID of the call to be transfered. + + + + + The callID of the target call. + + + - - - Start audio stream and play tone.. - - - - + + + Dual-Tone multi-frequency. Tell the core to play dialtones. A SIP INFO message is sent to notify the server. + + + + Unicode character for pressed key. + + + - - -

Sets the volume using a linear scale [0,100].

- Pulseaudio has its own mechanism to modify application volume. This method is enabled only if the ALSA API is used. - - - - The device: mic or speaker - - - - - The volume value (between 0 and 100) - - - + + + Start audio stream and play tone. + + + + - - -

Return the volume value of the given device on a linear scale [0,100].

- Only enabled if the ALSA API is used, Pulseaudio has its own mechanism to modify application volume. - - - - The device: mic or speaker - - - - - The volume value (between 0 and 100) - - - + + +

Sets the volume using a linear scale [0,100].

+ Pulseaudio has its own mechanism to modify application volume. This method is enabled only if the ALSA API is used. + + + + The device: mic or speaker + + + + + The volume value (between 0 and 100) + + + - - - -

Join two participants together to create a 3-way conference including the current client.

- The signal conferenceCreated is emitted on success. - - - - + + +

Return the volume value of the given device on a linear scale [0,100].

+ Only enabled if the ALSA API is used, Pulseaudio has its own mechanism to modify application volume. + + + + The device: mic or speaker + + + + + The volume value (between 0 and 100) + + + + + + + +

Join two participants together to create a 3-way conference including the current client.

+ The signal conferenceCreated is emitted on success. + + + + - - -

Create a conference from a list of participant

- The signal conferenceCreated is emitted on success. - - - + + +

Create a conference from a list of participants

+ The signal conferenceCreated is emitted on success. + + + - - - -

Join a new particiant to an existing conference.

- The signal conferenceChanged is emitted on success. - - - - The ID of the call to add to the conference - - - - - An existing conference ID - - - + + + +

Join a new particiant to an existing conference.

+ The signal conferenceChanged is emitted on success. + + + + The ID of the call to add to the conference + + + + + An existing conference ID + + + - - - -

As the core can handle multiple calls an conferences, it may happens that the client's user leave a conference to answer an incoming call or send new ones. This method is used to reintroduce SFLphone-client's user into the conference.

-

It put the current call on state HOLD or detach SFLphone-client's user from the another conference.

- - - - An existing conference ID - - - + + + +

As the core can handle multiple calls and conferences, it may happen that the client's user leaves a conference to answer an incoming call or to start new calls. This method is used to reintroduce SFLphone-client's user into the conference.

+

Its put the current call on HOLD or detaches SFLphone-client's user from the another conference.

+ + + + An existing conference ID + + + - - - - Detach the given call from the conference. If only one participant is left, the conference is deleted and the signal conferenceRemoved is emited. - - - - The call ID - - - + + + + Detach the given call from the conference. If only one participant is left, the conference is deleted and the signal conferenceRemoved is emited. + + + + The call ID + + + - - - - Join two conferences together. - - - - + + + + Join two conferences together. + + + + - - - - Returns a hashtable containing conference details. - - - - The conference ID - - - - - - A map containing the ID of the conferences - and their states: -
    -
  • ACTIVE_ATTACHED
    • -
  • ACTIVE_DETACHED
    • -
  • HOLD
    • -
- - - + + + + Returns a hashtable containing conference details. + + + + The conference ID + + + + + + A map containing the ID of the conferences + and their states: +
    +
  • ACTIVE_ATTACHED
    • +
  • ACTIVE_DETACHED
    • +
  • HOLD
    • +
+ + + - - - - Returns a list containing all active - conferences. - To update client status, one should - use getParticipantList - with provided conference IDs. - - - - The list of conferences. - - - + + + + Returns a list containing all active + conferences. + To update client status, one should + use getParticipantList + with provided conference IDs. + + + + The list of conferences. + + + - - - Start recording a call. - - - - The ID of the call to record. - - - + + + Start recording a call. + + + + The ID of the call to record. + + + - - - Tells whether or not a call is being recorded. - - - - The call ID. - - - - - Returns true is the call is being recorded. False otherwise. - - - + + + Tells whether or not a call is being recorded. + + + + The call ID. + + + + + Returns true is the call is being recorded. False otherwise. + + + - - - Once after starting recording for the first time, this signal is emited to - provide the recorded file path to client application. - - - - + + + Once after starting recording for the first time, this signal is emited to + provide the recorded file path to client application. + + + + - - - - - + + + + - - - Get all the details about a specific call. - - - - The call ID. - - - - - -

A map containing the call details:

-
    -
  • ACCOUNTID
    • -
  • PEER_NUMBER
    • -
  • PEER_NAME
    • -
  • DISPLAY_NAME
    • -
  • CALL_STATE
    • -
  • CALL_TYPE
    • -
- - - - - - Get the list of active calls. - To get the call details, iterate on the return value and call getCallDetails method. - - - - A list of call IDs. - - - + + + Get all the details about a specific call. + + + + The call ID. + + + + + +

A map containing the call details:

+
    +
  • ACCOUNTID
    • +
  • PEER_NUMBER
    • +
  • PEER_NAME
    • +
  • DISPLAY_NAME
    • +
  • CALL_STATE
    • +
  • CALL_TYPE
    • +
+ + + - - - Unused - - - - - - + + + Get the list of active calls. + To get the call details, iterate on the return value and call getCallDetails method. + + + + A list of call IDs. + + + - - - Unused - - - - + + + + - - - Send a text message to the specified call - - - - + + + Send a text message to the specified call + + + + - - -

Notify that a cell have been created.

-

The callID generated by the daemon must be stored by the clients in order to address other action for - this call. This signal is emitted when call have been created by the daemon itself.

- The client must subscribe to this signal to handle calls created by other clients - - - - The account ID of the calle. Clients must notify teh right account when receiving this signal. - - - - - A new call ID. - - - - - The sip uri this call is trying to reach - - - + + +

Notify that a call has been created.

+

The callID generated by the daemon must be stored by the clients in order to address other actions for + this call. This signal is emitted when call haves been created by the daemon itself.

+ The client must subscribe to this signal to handle calls created by other clients + + + + The account ID of the call. Clients must notify the right account when receiving this signal. + + + + + A new call ID. + + + + + The SIP URI this call is trying to reach. + + + - - -

Notify an incoming call.

-

The callID generated by the daemon must be stored by the clients in order to address other action for - this call. This signal is emitted when we receive a call from a remote peer

- The client must subscribe to this signal to handle incoming calls. - - - - The account ID of the callee. Clients must notify the right account when receiving this signal. - - - - - A new call ID. - - - - - The caller phone number. - - - + + +

Notify an incoming call.

+

The callID generated by the daemon must be stored by the clients in order to address other action for + this call. This signal is emitted when we receive a call from a remote peer

+ The client must subscribe to this signal to handle incoming calls. + + + + The account ID of the callee. Clients must notify the right account when receiving this signal. + + + + + A new call ID. + + + + + The caller phone number. + + + - - - Notify clients that a new text message has been received. - - - - - + + + Notify clients that a new text message has been received. + + + + + - - -

Notify of a change in a call state.

-

The client must subscribe to this signal.

- - - - The call ID. - - - - - The acceptable states are: -
    -
  • INCOMING: Initial state of incoming calls
    • -
  • RINGING: Initial state of received outgoing call
    • -
  • CURRENT: The normal active state of an answered call
    • -
  • HUNGUP: Notify that the call has been hungup by peer
    • -
  • BUSY
    • -
  • FAILURE: Error when processing a call
    • -
  • HOLD
    • -
  • UNHOLD_CURRENT
    • -
  • UNHOLD_RECORD
    • -
- - - + + +

Notify of a change in a call state.

+

The client must subscribe to this signal.

+ + + + The call ID. + + + + + The acceptable states are: +
    +
  • INCOMING: Initial state of incoming calls
    • +
  • RINGING: Initial state of received outgoing call
    • +
  • CURRENT: The normal active state of an answered call
    • +
  • HUNGUP: Notify that the call has been hungup by peer
    • +
  • BUSY
    • +
  • FAILURE: Error when processing a call
    • +
  • HOLD
    • +
  • UNHOLD_CURRENT
    • +
  • UNHOLD_RECORD
    • +
+ + + - - - - Notify of a change in the conferences state - - - - The conference ID. - - - - - The acceptable states are: -
    -
  • ACTIVE_ATTACHED: SFLphone user is - participating to this conference
    • -
  • ACTIVE_DETACHED: This situation can - occur if a call is received while - SFLphone user is participating to a - conference. In this case, one can leave - the conference by answering the - call. Other participants may continue - conferencing normally.
    • -
  • HOLD: Each call in this conference - is on state HOLD
    • -
- - - + + + + Notify of a change in the conferences state + + + + The conference ID. + + + + + The acceptable states are: +
    +
  • ACTIVE_ATTACHED: SFLphone user is + participating to this conference
    • +
  • ACTIVE_DETACHED: This situation can + occur if a call is received while + SFLphone user is participating to a + conference. In this case, one can leave + the conference by answering the + call. Other participants may continue + conferencing normally.
    • +
  • HOLD: Each call in this conference + is on state HOLD
    • +
+ + + - - - - Get the call IDs of every participant to a given conference. The client should keep and update the list of participant. - - - - The conference ID. - - - - - The list of the call IDs. - - - + + + + Get the call IDs of every participant to a given conference. The client should keep and update the list of participants. + + + + The conference ID. + + + + + The list of the call IDs. + + + - - - - Emited when a new conference is created. SFLphone-client is reponsible to store the confID and call getParticipantList to update the display. - - - - A new conference ID. - - - + + + + Emited when a new conference is created. SFLphone-client is reponsible for storing the confID and call getParticipantList to update the display. + + + + A new conference ID. + + + - - - - Emited when a new conference is remove. SFLphone-client should have kept a list of current participant in order to display modification. - - - - The conference ID. - - - + + + + Emited when a new conference is remove. SFLphone-client should have kept a list of current participant in order to display modification. + + + + The conference ID. + + + - - - - Hold on every calls participating to this conference. - - - - The conference ID. - - - + + + + Hold every call which is participating in this conference. + + + + The conference ID. + + + - - - - Hold off every calls participating to this conference. - - - - The conference ID. - - - - - - - - - - - + + + + Hold off every call participating in this conference. + + + + The conference ID. + + + - - - - - + + + + + - - -

Call state changed, SFLphone received a notification - from registrar concerning this call.

- - - - The call ID - - - - Description string - - - - The SIP or IAX2 message code - - + + + + + - - -

Account state changed, SFLphone received a notification - from registrar.

- - - - The account ID - - - - - Description string - - - - - The SIP or IAX2 message code - - - + + +

Call state changed, SFLphone received a notification + from registrar concerning this call.

+ + + + The call ID + + + + Description string + + + + The SIP or IAX2 message code + + - - - Notify the clients of the voicemail number for a specific account, if applicable. - - - - The account ID. - - - - - The number of waiting messages. - - - + + +

Account state changed, SFLphone received a notification + from registrar.

+ + + + The account ID + + + + + Description string + + + + + The SIP or IAX2 message code + + + - - -

Notify clients of a volume level - change.

-

This signal occurs only if ALSA is - enabled since Pulseaudio streams are - managed externally.

- - - - The device: mic or speaker - - - - - The new volume value - - - + + + Notify the clients of the voicemail number for a specific account, if applicable. + + + + The account ID. + + + + + The number of waiting messages. + + + - - -

Transfer has been successfully - processed. Client should remove transfered - call from call list as it is no longer - accessible in SFLphone-daemon.

- - + + +

Notify clients of a volume level change.

+

This signal occurs only if ALSA is enabled since Pulseaudio streams are managed externally.

+ + + + The device: mic or speaker + + + + + The new volume value + + + - - -

Transfer operation failed. Corespondin - call is no longer accessible in - SFLphone-daemon.

- - + + +

Transfer has been successfully + processed. Client should remove transfered + call from call list as it is no longer + accessible in SFLphone-daemon.

+ + - - - -

Signal sent on SDES session success. Media transmission is encripted - for this call only. It does not apply for a - conference.

-

A conference can be considered to be secured if and only if each - participant is secured.

- - - + + +

Transfer operation failed. Corespondin + call is no longer accessible in + SFLphone-daemon.

+ + - - - -

Sinal sent to notify that SDES session - failed.

-

Media transmission is not encrypted.

- - - + + + +

Signal sent on SDES session success. Media transmission is encripted + for this call only. It does not apply for a conference.

+

A conference can be considered to be secured if and only if each + participant is secured.

+ + + - - - - - - - - + + + +

Sinal sent to notify that SDES session failed.

+

Media transmission is not encrypted.

+ + + - - - - - - + + + + + + - - - - - - + + + + - - - - - - - - + + + + - - - - - - + + + + + + - - - - - - - - - + + + + - - - - - - + + + + + + + - - - - - - + + + + - - - - - - + + + + - - - - - - + + + + - - - - - - - + + + + - - - - - - - + + + + + - + + + + + + + diff --git a/kde/src/dbus/configurationmanager-introspec.xml b/kde/src/dbus/configurationmanager-introspec.xml index d86ffc2b6..5f30f2526 100755 --- a/kde/src/dbus/configurationmanager-introspec.xml +++ b/kde/src/dbus/configurationmanager-introspec.xml @@ -1,107 +1,107 @@ - + - - Used to handle the configuration stuff: accounts settings, account registration, user preferences, ... - + + Used to handle the configuration stuff: accounts settings, account registration, user preferences, ... + - - - Get all parameters of the specified account. - - - - The account ID - - - - - - The available keys / parameters are: -
    -
  • CONFIG_ACCOUNT_ENABLE: True or False (Default: True)
    • -
  • CONFIG_ACCOUNT_RESOLVE_ONCE
    • -
  • CONFIG_ACCOUNT_TYPE: SIP or IAX2 (Default: SIP)
    • -
  • HOSTNAME: The IP adress or hostname of the registrar
    • -
  • USERNAME: The username (or extension) of the account
    • -
  • PASSWORD: The password associated to the account
    • -
  • REALM
    • -
  • CONFIG_ACCOUNT_MAILBOX: Number to dial to access the voicemail box
    • -
  • CONFIG_ACCOUNT_REGISTRATION_EXPIRE: SIP header expiration value (Default: 1600)
    • -
  • LOCAL_INTERFACE: The network interface (Default: eth0)
    • -
  • PUBLISHED_SAMEAS_LOCAL: If False, the published address equals the local address. This is the default.
    • -
  • PUBLISHED_ADDRESS: The SIP published address
    • -
  • LOCAL_PORT: The SIP listening port (Default: 5060)
    • -
  • PUBLISHED_PORT: The SIP published port
    • -
  • DISPLAY_NAMEL: The display name
    • -
  • STUN_ENABLE: True or False (Default: False)
    • -
  • STUN_SERVER: The STUN server address
    • -
  • REGISTRATION_STATUS: The account registration status. Should be Registered to make calls.
    • -
  • REGISTRATION_STATE_CODE
    • -
  • REGISTRATION_STATE_DESCRIPTION
    • -
  • SRTP_KEY_EXCHANGE
    • -
  • SRTP_ENABLE: Whether or not voice communication are encrypted - True or False (Default: False)
    • -
  • SRTP_RTP_FALLBACK
    • -
  • ZRTP_DISPLAY_SAS
    • -
  • ZRTP_DISPLAY_SAS_ONCE
    • -
  • ZRTP_HELLO_HASH
    • -
  • ZRTP_NOT_SUPP_WARNING
    • -
  • TLS_LISTENER_PORT: TLS listening port (Default: 5061)
    • -
  • TLS_ENABLE: Whether or not signalling is encrypted - True or False (Default: False)
    • -
  • TLS_CA_LIST_FILE
    • -
  • TLS_CERTIFICATE_FILE
    • -
  • TLS_PRIVATE_KEY_FILE
    • -
  • TLS_METHOD
    • -
  • TLS_CIPHERS
    • -
  • TLS_SERVER_NAME
    • -
  • TLS_VERIFY_SERVER
    • -
  • TLS_VERIFY_CLIENT
    • -
  • TLS_REQUIRE_CLIENT_CERTIFICATE
    • -
  • TLS_NEGOTIATION_TIMEOUT_SEC
    • -
  • TLS_NEGOTIATION_TIMEOUT_MSEC
    • -
- - - + + + Get all parameters of the specified account. + + + + The account ID + + + + + + The available keys / parameters are: +
    +
  • CONFIG_ACCOUNT_ENABLE: True or False (Default: True)
    • +
  • CONFIG_ACCOUNT_RESOLVE_ONCE
    • +
  • CONFIG_ACCOUNT_TYPE: SIP or IAX2 (Default: SIP)
    • +
  • HOSTNAME: The IP adress or hostname of the registrar
    • +
  • USERNAME: The username (or extension) of the account
    • +
  • PASSWORD: The password associated to the account
    • +
  • REALM
    • +
  • CONFIG_ACCOUNT_MAILBOX: Number to dial to access the voicemail box
    • +
  • CONFIG_ACCOUNT_REGISTRATION_EXPIRE: SIP header expiration value (Default: 1600)
    • +
  • LOCAL_INTERFACE: The network interface (Default: eth0)
    • +
  • PUBLISHED_SAMEAS_LOCAL: If False, the published address equals the local address. This is the default.
    • +
  • PUBLISHED_ADDRESS: The SIP published address
    • +
  • LOCAL_PORT: The SIP listening port (Default: 5060)
    • +
  • PUBLISHED_PORT: The SIP published port
    • +
  • DISPLAY_NAMEL: The display name
    • +
  • STUN_ENABLE: True or False (Default: False)
    • +
  • STUN_SERVER: The STUN server address
    • +
  • REGISTRATION_STATUS: The account registration status. Should be Registered to make calls.
    • +
  • REGISTRATION_STATE_CODE
    • +
  • REGISTRATION_STATE_DESCRIPTION
    • +
  • SRTP_KEY_EXCHANGE
    • +
  • SRTP_ENABLE: Whether or not voice communication are encrypted - True or False (Default: False)
    • +
  • SRTP_RTP_FALLBACK
    • +
  • ZRTP_DISPLAY_SAS
    • +
  • ZRTP_DISPLAY_SAS_ONCE
    • +
  • ZRTP_HELLO_HASH
    • +
  • ZRTP_NOT_SUPP_WARNING
    • +
  • TLS_LISTENER_PORT: TLS listening port (Default: 5061)
    • +
  • TLS_ENABLE: Whether or not signalling is encrypted - True or False (Default: False)
    • +
  • TLS_CA_LIST_FILE
    • +
  • TLS_CERTIFICATE_FILE
    • +
  • TLS_PRIVATE_KEY_FILE
    • +
  • TLS_METHOD
    • +
  • TLS_CIPHERS
    • +
  • TLS_SERVER_NAME
    • +
  • TLS_VERIFY_SERVER
    • +
  • TLS_VERIFY_CLIENT
    • +
  • TLS_REQUIRE_CLIENT_CERTIFICATE
    • +
  • TLS_NEGOTIATION_TIMEOUT_SEC
    • +
  • TLS_NEGOTIATION_TIMEOUT_MSEC
    • +
+ + + - - - Send new account parameters, or account parameters changes, to the core. The hash table is not required to be complete, only the updated parameters may be specified. - Account settings are written to the configuration file when sflphone properly quits. - After calling this method, the core will emit the signal accountsChanged with the updated data. The client must subscribe to this signal and use it to update its internal data structure. - - - - - - - - - - - + + + Send new account parameters, or account parameters changes, to the core. The hash table is not required to be complete, only the updated parameters may be specified. + Account settings are written to the configuration file when sflphone properly quits. + After calling this method, the core will emit the signal accountsChanged with the updated data. The client must subscribe to this signal and use it to update its internal data structure. + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - + Get configuration settings of the IP2IP_PROFILE. They are sligthly different from account settings since no VoIP accounts are involved. - + Available parameters are: @@ -134,584 +134,584 @@ - - - - - - - - - - - - - + + + + + + + + + + + + + - - - Add a new account. When created, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. - If no details are specified, the default parameters are used. - The core tries to register the account as soon it is created. - - - - - The new account settings - - - - - A new account ID - - - + + + Add a new account. When created, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. + If no details are specified, the default parameters are used. + The core tries to register the account as soon it is created. + + + + + The new account settings + + + + + A new account ID + + + - - - Update the accounts order. - When placing a call, the first registered account in the list is used. - - - - An ordered list of account IDs, delimited by '/' - - - + + + Update the accounts order. + When placing a call, the first registered account in the list is used. + + + + An ordered list of account IDs, delimited by '/' + + + - - - Remove an existing account. When removed, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. - - - - The account to remove, identified by its ID - - - + + + Remove an existing account. When removed, the signal accountsChanged is emitted. The clients must then call getAccountList to update their internal data structure. + + + + The account to remove, identified by its ID + + + - - - Get a list of all created accounts, as stored by the core. - - - - - A list of account IDs - - - + + + Get a list of all created accounts, as stored by the core. + + + + + A list of account IDs + + + - - - Send account registration (REGISTER) to the registrar. - - the account if expire=1, unregister if expire=0. + + + Send account registration (REGISTER) to the registrar. + + the account if expire=1, unregister if expire=0. - @param[in] input accountID - --> - - - The account ID - - - - -

To register, expire must be 1.

-

To un-register, expire must be 0.

- - - + @param[in] input accountID + --> + + + The account ID + + + + +

To register, expire must be 1.

+

To un-register, expire must be 0.

+ + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - + - + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - - - - - - - - - - - - - + + + + + + + + + + + + + - + - - - - - - - - - + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - + + + + + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - + + + + + + + - - - - + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - + + + + + + - - - - - - + + + + + + - - - - - - - - + + + + + + + + - - - - + + + + - + - - - - - - - - + + + + + + + + - - - - + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - + + + + + + + + + + - - - - - - - - - + + + + + + + + + - + - + - + - - + + - - - - - - + + + + + + - - - - - - - - - - + + + + + + + + + + - + - - - - - - - + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - - - - + + + + + + + + + + + + - - - - - - - - - + + + + + + + + + - - - - - - - - - + + + + + + + + + - + - - - - - - - + + + + + + + - + - - - - - - - + + + + + + + - +