Functions | |
| int | am_session_start (const char *identity, const char *url, const char *proxy, const char *outbound_proxy) |
| int | am_session_start_text_conversation (const char *identity, const char *url, const char *proxy, const char *outbound_proxy) |
| int | am_session_get_text (int did, MST140Block *_blk) |
| int | am_session_send_text (int did, MST140Block *_blk) |
| void | am_free_t140block (MST140Block *_blk) |
| int | am_session_hold (int did, const char *wav_file) |
| int | am_session_off_hold (int did) |
| int | am_session_inactive (int did) |
| int | am_session_inactive_0_0_0_0 (int did) |
| int | am_session_mute (int did) |
| int | am_session_unmute (int did) |
| int | am_session_add_in_conference (const char *identity, const char *url, const char *proxy, const char *outbound_proxy, const char *conf_name) |
| int | am_session_turn_into_conference (int did, const char *conf_name) |
| int | am_session_answer (int tid, int did, int code, int enable_audio) |
| int | am_session_redirect (int tid, int did, int code, const char *url) |
| int | am_session_stop (int cid, int did, int code) |
| int | am_session_refer (int did, const char *url, const char *referred_by) |
| int | am_session_get_referto (int did, char *refer_to, size_t refer_to_len) |
| int | am_session_find_by_replaces (osip_message_t *request) |
| int | am_session_answer_request (int tid, int did, int code) |
| int | am_session_send_request (int did, const char *method, const char *content_type, const char *body, int size) |
| int | am_session_play_file (int did, const char *wav_file) |
| int | am_session_send_rtp_dtmf (int did, char dtmf_number) |
| int | am_session_send_dtmf (int did, char dtmf_number) |
| int | am_session_get_dtmf_event (int did, struct am_dtmf_event *dtmf_event) |
| int | am_session_get_audio_remote (int did, char *remote_info) |
| int | am_session_get_audio_statistics (int did, struct am_audio_stats *audio_stats) |
| int | am_session_modify_bitrate (int did, const char *preferred_codec, int compress_more) |
| int | am_session_record (int did, const char *recfile) |
| int | am_session_stop_record (int did) |
| int | am_session_add_t140 (int did) |
| int | am_session_add_external_rtpdata (int did, MSFilter *external_rtpdata) |
| int am_session_start | ( | const char * | identity, | |
| const char * | url, | |||
| const char * | proxy, | |||
| const char * | outbound_proxy | |||
| ) |
Configure amsip to start a SIP session.
| identity | Caller SIP identity | |
| url | Callee SIP identity | |
| proxy | Set the proxy server | |
| outbound_proxy | OutBound Proxy |
| int am_session_start_text_conversation | ( | const char * | identity, | |
| const char * | url, | |||
| const char * | proxy, | |||
| const char * | outbound_proxy | |||
| ) |
Configure amsip to start a SIP session with text.
| identity | Caller SIP identity | |
| url | Callee SIP identity | |
| proxy | Set the proxy server | |
| outbound_proxy | OutBound Proxy |
| int am_session_get_text | ( | int | did, | |
| MST140Block * | _blk | |||
| ) |
Get UDP/RTP text from remote party.
| did | Session identifier. | |
| _blk | Block that contains text data. |
| int am_session_send_text | ( | int | did, | |
| MST140Block * | _blk | |||
| ) |
Send UDP/RTP text from remote party.
| did | Session identifier. | |
| _blk | Block that contains text data. |
| void am_free_t140block | ( | MST140Block * | _blk | ) |
Release allocated string inside MST140Block.
| _blk | Block that contains text data. |
| int am_session_hold | ( | int | did, | |
| const char * | wav_file | |||
| ) |
Configure amsip to put on hold a SIP session. (send wav_file)
| did | Session identifier. | |
| wav_file | Wav file to play. |
| int am_session_off_hold | ( | int | did | ) |
Configure amsip to put off hold a SIP session.
| did | Session identifier. |
| int am_session_inactive | ( | int | did | ) |
Configure amsip to make streams inactive.
| did | Session identifier. |
| int am_session_inactive_0_0_0_0 | ( | int | did | ) |
Configure amsip to make streams inactive. Note: put ip=0_0_0_0 and port=0 for the audio SDP media block. Note2: THIS IS NOT POSSIBLE TO RETREIVE BACK THIS CALL. THIS METHOD MUST ONLY BE USED BEFORE MAKING A CALL TRANSFER WITH SOME UNCOMPLIANT SIP-PBX.
| did | Session identifier. |
| int am_session_mute | ( | int | did | ) |
Configure amsip to mute a SIP session. (send silence)
| did | Session identifier. |
| int am_session_unmute | ( | int | did | ) |
Configure amsip to unmute a SIP session.
| did | Session identifier. |
| int am_session_add_in_conference | ( | const char * | identity, | |
| const char * | url, | |||
| const char * | proxy, | |||
| const char * | outbound_proxy, | |||
| const char * | conf_name | |||
| ) |
Configure amsip to start a SIP session and add user in an existing conference.
| identity | Caller SIP identity | |
| url | Callee SIP identity | |
| proxy | Set the proxy server | |
| outbound_proxy | OutBound Proxy | |
| conf_name | Conference name |
| int am_session_turn_into_conference | ( | int | did, | |
| const char * | conf_name | |||
| ) |
Configure amsip to add a session in a conference.
| did | Session identifier. | |
| conf_name | Conference identifier. |
| int am_session_answer | ( | int | tid, | |
| int | did, | |||
| int | code, | |||
| int | enable_audio | |||
| ) |
Configure amsip to establish a SIP session.
| tid | Transaction identifier. | |
| did | Session identifier. | |
| code | Code to use. | |
| enable_audio | enable audio in answer. |
| int am_session_redirect | ( | int | tid, | |
| int | did, | |||
| int | code, | |||
| const char * | url | |||
| ) |
Configure amsip to redirect a SIP session.
| tid | Transaction identifier. | |
| did | Session identifier. | |
| code | Code to use. | |
| url | SIP (or other url) address for redirection. |
| int am_session_stop | ( | int | cid, | |
| int | did, | |||
| int | code | |||
| ) |
Configure amsip to establish a SIP session.
| cid | dialog identifier. | |
| did | Session identifier. | |
| code | Code to use (if answer needed). |
| int am_session_refer | ( | int | did, | |
| const char * | url, | |||
| const char * | referred_by | |||
| ) |
Configure amsip to transfer a SIP session.
| did | Session identifier. | |
| url | SIP (or other url) address for redirection. |
| int am_session_get_referto | ( | int | did, | |
| char * | refer_to, | |||
| size_t | refer_to_len | |||
| ) |
Add Replaces parameter to refer-to url.
| did | Session identifier. | |
| refer_to | String to carry refer_to. | |
| refer_to_len | Size of refer_to string. |
| int am_session_find_by_replaces | ( | osip_message_t * | request | ) |
Find the call that relates to the Replaces header.
| request | requet containging a replace header. |
| int am_session_answer_request | ( | int | tid, | |
| int | did, | |||
| int | code | |||
| ) |
Configure amsip to establish a SIP session.
| tid | Transaction identifier. | |
| did | Session identifier. | |
| code | Code to use. |
| int am_session_send_request | ( | int | did, | |
| const char * | method, | |||
| const char * | content_type, | |||
| const char * | body, | |||
| int | size | |||
| ) |
Send a request during a session
| did | Session identifier. | |
| method | SIP method (e.g. INVITE) | |
| content_type | Content-type of the body | |
| body | Attached content | |
| size | Size of body |
| int am_session_play_file | ( | int | did, | |
| const char * | wav_file | |||
| ) |
Configure amsip to play a file in an outgoing RTP session.
| did | Session identifier. | |
| wav_file | File to play. |
| int am_session_send_rtp_dtmf | ( | int | did, | |
| char | dtmf_number | |||
| ) |
Configure amsip to send a RTP telephone-event in an outgoing RTP session.
| did | Session identifier. | |
| dtmf_number | DTMF to send. |
| int am_session_send_dtmf | ( | int | did, | |
| char | dtmf_number | |||
| ) |
Configure amsip to send an INFO with dtmf attachement.
| did | Session identifier. | |
| dtmf_number | DTMF to send. |
| int am_session_get_dtmf_event | ( | int | did, | |
| struct am_dtmf_event * | dtmf_event | |||
| ) |
Get dtmf from RTP session.
| did | Session identifier. | |
| dtmf_event | dtmf_event structure. |
| int am_session_get_audio_remote | ( | int | did, | |
| char * | remote_info | |||
| ) |
Find out the destination IP/PORT for audio stream. return >0 remote SDP ip/port used return ==0 remote SDP candidate ip/port used (probably for direct comunication) return <0 undefined error.
| did | Session identifier. | |
| remote_info | string of size 256 to get the remote IP/PORT for audio stream. |
| int am_session_get_audio_statistics | ( | int | did, | |
| struct am_audio_stats * | audio_stats | |||
| ) |
Get statistics for RTP audio stream.
| did | Session identifier. | |
| audio_stats | struct to hold statistics. |
| int am_session_modify_bitrate | ( | int | did, | |
| const char * | preferred_codec, | |||
| int | compress_more | |||
| ) |
Get statistics for RTP audio stream.
| did | Session identifier. | |
| preferred_codec | Preferred codec to use. | |
| compress_more | compress_more. (0 means you want to use more bandwidth) |
| int am_session_record | ( | int | did, | |
| const char * | recfile | |||
| ) |
Record both audio stream (incoming+outgoing).
| did | Session identifier. | |
| recfile | File Name for recording. |
| int am_session_stop_record | ( | int | did | ) |
Stop recording the call.
| did | Session identifier. |
| int am_session_add_t140 | ( | int | did | ) |
Add t140 chat session in the call.
| did | Session identifier. |
| int am_session_add_external_rtpdata | ( | int | did, | |
| MSFilter * | external_rtpdata | |||
| ) |
Add new source for sending RTP data.
| did | Session identifier. | |
| external_rtpdata | External filter. |
1.5.4