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sqozz b4b1ba31dc Fix status parsing 8 months ago
README.md Add version function 8 months ago
pylinphone.py Fix status parsing 8 months ago

README.md

pylinphone

Currently implemented Features:

  • register a new account (register)
  • check status for an registered account (register-status and register-info)
  • event-queue polling and onEvent functions (pop-event)
  • create an outgoing call (call)
  • answer an incoming call (accept)
  • terminate a running call (terminate)
  • mute the call (mute-call)
  • pause the call (call-pause)
  • resume the call (call-resume)
  • play dtmf tones (dtmf)
  • get call status (call-status)
  • get call stats (call-stats)
  • unregister an account (unregister)
  • get daemon version (version)

Features supported by the unix socket (linphone deamon):

adaptive-jitter-compensation [audio|video] [enable|disable]
answer <call_id>
audio-codec-disable <payload_type_number>|<mime_type>|ALL
audio-codec-enable <payload_type_number>|<mime_type>|ALL
audio-codec-get <payload_type_number>|<mime_type>
audio-codec-move <payload_type_number>|<mime_type> <index>
audio-codec-set <payload_type_number>|<mime_type> <property> <value>
audio-stream-start <remote_ip> <remote_port> <payload_type_number>
audio-stream-stats <stream_id>
audio-stream-stop <stream_id>
auth-infos-clear <auth_infos_id>|ALL
autovideo on|off
call <sip_address> [--early-media]
call-mute 0|1
call-pause [<call_id>]
call-resume [<call_id>]
call-stats [<call_id>]
call-status [<call_id>]
call-transfer <call_to_transfer_id> <call_to_transfer_to_id>|<sip_url_to_transfer_to>
cn [enable|disable]
conference add|rm|leave|enter <call_id>
config-get <section> <key>
config-set <section> <key> <value>
contact <sip_address> or contact <username> <hostname>
dtmf <digit>
firewall-policy [none|nat|stun|ice|upnp] [<address>]
help [<command>]
incall-player-pause [<call_id>]
incall-player-resume [<call_id>]
incall-player-start <filename> [<call_id>]
incall-player-stop [<call_id>]
ipv6 [enable|disable]
jitter-buffer [audio|video] [size <milliseconds>]
jitter-buffer-reset call|stream <id> [audio|video]
media-encryption [none|srtp|zrtp]
message <sip_address> <text>
msfilter-add-fmtp call|stream <id> <fmtp>
netsim [enable|disable|parameters] [<parameters>]
play-wav <filename>
pop-event
port [sip|audio|video] [<port>] [udp|tcp|tls]
ptime [up|down] [<ms>]
quit
register <identity> <proxy_address> [<password>] [<userid>] [<realm>] [<parameter>]
register-info <register_id>|ALL
register-status <register_id>|ALL
terminate [<call_id>]
unregister <register_id>|ALL
version
video [call_id]
videosource cam|dummy [<call_id>]