提交 1207c64a authored 作者: Giovanni Maruzzelli's avatar Giovanni Maruzzelli

skypiax: cleaning

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@16829 d0543943-73ff-0310-b7d9-9358b9ac24b2
上级 305821a9
......@@ -215,10 +215,10 @@ struct private_object {
#endif
struct SkypiaxHandles SkypiaxHandles;
int interface_state; /*!< \brief 'state' of the interface (channel) */
char language[80]; /*!< \brief default Asterisk dialplan language for this interface */
char exten[80]; /*!< \brief default Asterisk dialplan extension for this interface */
int skypiax_sound_rate; /*!< \brief rate of the sound device, in Hz, eg: 8000 */
int interface_state;
char language[80];
char exten[80];
int skypiax_sound_rate;
char callid_name[50];
char callid_number[50];
double playback_boost;
......@@ -229,17 +229,17 @@ struct private_object {
char skype_friends[4096];
char skype_fullname[512];
char skype_displayname[512];
int skype_callflow; /*!< \brief 'callflow' of the skype interface (as opposed to phone interface) */
int skype; /*!< \brief config flag, bool, Skype support on this interface (0 if false, -1 if true) */
int skype_callflow;
int skype;
int control_to_send;
#ifdef WIN32
switch_file_t *audiopipe_srv[2];
switch_file_t *audiopipe_cli[2];
switch_file_t *skypiax_sound_capt_fd; /*!< \brief file descriptor for sound capture dev */
switch_file_t *skypiax_sound_capt_fd;
#else /* WIN32 */
int audiopipe_srv[2];
int audiopipe_cli[2];
int skypiax_sound_capt_fd; /*!< \brief file descriptor for sound capture dev */
int skypiax_sound_capt_fd;
#endif /* WIN32 */
switch_thread_t *tcp_srv_thread;
switch_thread_t *tcp_cli_thread;
......@@ -254,14 +254,6 @@ struct private_object {
switch_mutex_t *mutex_audio_srv;
int flag_audio_srv;
//int phonebook_listing;
//int phonebook_querying;
//int phonebook_listing_received_calls;
//int phonebook_first_entry;
//int phonebook_last_entry;
//int phonebook_number_lenght;
//int phonebook_text_lenght;
FILE *phonebook_writing_fp;
int skypiax_dir_entry_extension_prefix;
char skype_user[256];
......
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